aboutsummaryrefslogtreecommitdiff
path: root/sound/soc/codecs
diff options
context:
space:
mode:
Diffstat (limited to 'sound/soc/codecs')
-rw-r--r--sound/soc/codecs/Kconfig22
-rw-r--r--sound/soc/codecs/Makefile8
-rw-r--r--sound/soc/codecs/ac97.c31
-rw-r--r--sound/soc/codecs/ac97.h2
-rw-r--r--sound/soc/codecs/ak4535.c696
-rw-r--r--sound/soc/codecs/ak4535.h46
-rw-r--r--sound/soc/codecs/cs4270.c8
-rw-r--r--sound/soc/codecs/cs4270.h2
-rw-r--r--sound/soc/codecs/tlv320aic3x.c384
-rw-r--r--sound/soc/codecs/tlv320aic3x.h55
-rw-r--r--sound/soc/codecs/uda1380.c852
-rw-r--r--sound/soc/codecs/uda1380.h89
-rw-r--r--sound/soc/codecs/wm8510.c817
-rw-r--r--sound/soc/codecs/wm8510.h103
-rw-r--r--sound/soc/codecs/wm8731.c79
-rw-r--r--sound/soc/codecs/wm8731.h2
-rw-r--r--sound/soc/codecs/wm8750.c87
-rw-r--r--sound/soc/codecs/wm8750.h2
-rw-r--r--sound/soc/codecs/wm8753.c183
-rw-r--r--sound/soc/codecs/wm8753.h2
-rw-r--r--sound/soc/codecs/wm8990.c1626
-rw-r--r--sound/soc/codecs/wm8990.h832
-rw-r--r--sound/soc/codecs/wm9712.c53
-rw-r--r--sound/soc/codecs/wm9712.h2
-rw-r--r--sound/soc/codecs/wm9713.c79
-rw-r--r--sound/soc/codecs/wm9713.h2
26 files changed, 5596 insertions, 468 deletions
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 3903ab7dfa4..1db04a28a53 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -1,31 +1,37 @@
config SND_SOC_AC97_CODEC
tristate
- depends on SND_SOC
+ select SND_AC97_CODEC
+
+config SND_SOC_AK4535
+ tristate
+
+config SND_SOC_UDA1380
+ tristate
+
+config SND_SOC_WM8510
+ tristate
config SND_SOC_WM8731
tristate
- depends on SND_SOC
config SND_SOC_WM8750
tristate
- depends on SND_SOC
config SND_SOC_WM8753
tristate
- depends on SND_SOC
+
+config SND_SOC_WM8990
+ tristate
config SND_SOC_WM9712
tristate
- depends on SND_SOC
config SND_SOC_WM9713
tristate
- depends on SND_SOC
# Cirrus Logic CS4270 Codec
config SND_SOC_CS4270
tristate
- depends on SND_SOC
# Cirrus Logic CS4270 Codec Hardware Mute Support
# Select if you have external muting circuitry attached to your CS4270.
@@ -43,4 +49,4 @@ config SND_SOC_CS4270_VD33_ERRATA
config SND_SOC_TLV320AIC3X
tristate
- depends on SND_SOC && I2C
+ depends on I2C
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 4e1314c9d3e..d7b97abcf72 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -1,16 +1,24 @@
snd-soc-ac97-objs := ac97.o
+snd-soc-ak4535-objs := ak4535.o
+snd-soc-uda1380-objs := uda1380.o
+snd-soc-wm8510-objs := wm8510.o
snd-soc-wm8731-objs := wm8731.o
snd-soc-wm8750-objs := wm8750.o
snd-soc-wm8753-objs := wm8753.o
+snd-soc-wm8990-objs := wm8990.o
snd-soc-wm9712-objs := wm9712.o
snd-soc-wm9713-objs := wm9713.o
snd-soc-cs4270-objs := cs4270.o
snd-soc-tlv320aic3x-objs := tlv320aic3x.o
obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o
+obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o
+obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o
+obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o
obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o
obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o
obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o
+obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o
obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o
obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o
obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index 2a1ffe39690..61fd96ca7bc 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -10,9 +10,6 @@
* Free Software Foundation; either version 2 of the License, or (at your
* option) any later version.
*
- * Revision history
- * 17th Oct 2005 Initial version.
- *
* Generic AC97 support.
*/
@@ -24,6 +21,7 @@
#include <sound/ac97_codec.h>
#include <sound/initval.h>
#include <sound/soc.h>
+#include "ac97.h"
#define AC97_VERSION "0.6"
@@ -43,7 +41,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream)
SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\
SNDRV_PCM_RATE_48000)
-struct snd_soc_codec_dai ac97_dai = {
+struct snd_soc_dai ac97_dai = {
.name = "AC97 HiFi",
.type = SND_SOC_DAI_AC97,
.playback = {
@@ -146,9 +144,34 @@ static int ac97_soc_remove(struct platform_device *pdev)
return 0;
}
+#ifdef CONFIG_PM
+static int ac97_soc_suspend(struct platform_device *pdev, pm_message_t msg)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+ snd_ac97_suspend(socdev->codec->ac97);
+
+ return 0;
+}
+
+static int ac97_soc_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+ snd_ac97_resume(socdev->codec->ac97);
+
+ return 0;
+}
+#else
+#define ac97_soc_suspend NULL
+#define ac97_soc_resume NULL
+#endif
+
struct snd_soc_codec_device soc_codec_dev_ac97 = {
.probe = ac97_soc_probe,
.remove = ac97_soc_remove,
+ .suspend = ac97_soc_suspend,
+ .resume = ac97_soc_resume,
};
EXPORT_SYMBOL_GPL(soc_codec_dev_ac97);
diff --git a/sound/soc/codecs/ac97.h b/sound/soc/codecs/ac97.h
index 2bf6d69fd06..281aa42e2bb 100644
--- a/sound/soc/codecs/ac97.h
+++ b/sound/soc/codecs/ac97.h
@@ -14,6 +14,6 @@
#define __LINUX_SND_SOC_AC97_H
extern struct snd_soc_codec_device soc_codec_dev_ac97;
-extern struct snd_soc_codec_dai ac97_dai;
+extern struct snd_soc_dai ac97_dai;
#endif
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
new file mode 100644
index 00000000000..b26003c4f3e
--- /dev/null
+++ b/sound/soc/codecs/ak4535.c
@@ -0,0 +1,696 @@
+/*
+ * ak4535.c -- AK4535 ALSA Soc Audio driver
+ *
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Author: Richard Purdie <richard@openedhand.com>
+ *
+ * Based on wm8753.c by Liam Girdwood
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <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 "ak4535.h"
+
+#define AUDIO_NAME "ak4535"
+#define AK4535_VERSION "0.3"
+
+struct snd_soc_codec_device soc_codec_dev_ak4535;
+
+/* codec private data */
+struct ak4535_priv {
+ unsigned int sysclk;
+};
+
+/*
+ * ak4535 register cache
+ */
+static const u16 ak4535_reg[AK4535_CACHEREGNUM] = {
+ 0x0000, 0x0080, 0x0000, 0x0003,
+ 0x0002, 0x0000, 0x0011, 0x0001,
+ 0x0000, 0x0040, 0x0036, 0x0010,
+ 0x0000, 0x0000, 0x0057, 0x0000,
+};
+
+/*
+ * read ak4535 register cache
+ */
+static inline unsigned int ak4535_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg >= AK4535_CACHEREGNUM)
+ return -1;
+ return cache[reg];
+}
+
+static inline unsigned int ak4535_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u8 data;
+ data = reg;
+
+ if (codec->hw_write(codec->control_data, &data, 1) != 1)
+ return -EIO;
+
+ if (codec->hw_read(codec->control_data, &data, 1) != 1)
+ return -EIO;
+
+ return data;
+};
+
+/*
+ * write ak4535 register cache
+ */
+static inline void ak4535_write_reg_cache(struct snd_soc_codec *codec,
+ u16 reg, unsigned int value)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg >= AK4535_CACHEREGNUM)
+ return;
+ cache[reg] = value;
+}
+
+/*
+ * write to the AK4535 register space
+ */
+static int ak4535_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[2];
+
+ /* data is
+ * D15..D8 AK4535 register offset
+ * D7...D0 register data
+ */
+ data[0] = reg & 0xff;
+ data[1] = value & 0xff;
+
+ ak4535_write_reg_cache(codec, reg, value);
+ if (codec->hw_write(codec->control_data, data, 2) == 2)
+ return 0;
+ else
+ return -EIO;
+}
+
+static int ak4535_sync(struct snd_soc_codec *codec)
+{
+ u16 *cache = codec->reg_cache;
+ int i, r = 0;
+
+ for (i = 0; i < AK4535_CACHEREGNUM; i++)
+ r |= ak4535_write(codec, i, cache[i]);
+
+ return r;
+};
+
+static const char *ak4535_mono_gain[] = {"+6dB", "-17dB"};
+static const char *ak4535_mono_out[] = {"(L + R)/2", "Hi-Z"};
+static const char *ak4535_hp_out[] = {"Stereo", "Mono"};
+static const char *ak4535_deemp[] = {"44.1kHz", "Off", "48kHz", "32kHz"};
+static const char *ak4535_mic_select[] = {"Internal", "External"};
+
+static const struct soc_enum ak4535_enum[] = {
+ SOC_ENUM_SINGLE(AK4535_SIG1, 7, 2, ak4535_mono_gain),
+ SOC_ENUM_SINGLE(AK4535_SIG1, 6, 2, ak4535_mono_out),
+ SOC_ENUM_SINGLE(AK4535_MODE2, 2, 2, ak4535_hp_out),
+ SOC_ENUM_SINGLE(AK4535_DAC, 0, 4, ak4535_deemp),
+ SOC_ENUM_SINGLE(AK4535_MIC, 1, 2, ak4535_mic_select),
+};
+
+static const struct snd_kcontrol_new ak4535_snd_controls[] = {
+ SOC_SINGLE("ALC2 Switch", AK4535_SIG1, 1, 1, 0),
+ SOC_ENUM("Mono 1 Output", ak4535_enum[1]),
+ SOC_ENUM("Mono 1 Gain", ak4535_enum[0]),
+ SOC_ENUM("Headphone Output", ak4535_enum[2]),
+ SOC_ENUM("Playback Deemphasis", ak4535_enum[3]),
+ SOC_SINGLE("Bass Volume", AK4535_DAC, 2, 3, 0),
+ SOC_SINGLE("Mic Boost (+20dB) Switch", AK4535_MIC, 0, 1, 0),
+ SOC_ENUM("Mic Select", ak4535_enum[4]),
+ SOC_SINGLE("ALC Operation Time", AK4535_TIMER, 0, 3, 0),
+ SOC_SINGLE("ALC Recovery Time", AK4535_TIMER, 2, 3, 0),
+ SOC_SINGLE("ALC ZC Time", AK4535_TIMER, 4, 3, 0),
+ SOC_SINGLE("ALC 1 Switch", AK4535_ALC1, 5, 1, 0),
+ SOC_SINGLE("ALC 2 Switch", AK4535_ALC1, 6, 1, 0),
+ SOC_SINGLE("ALC Volume", AK4535_ALC2, 0, 127, 0),
+ SOC_SINGLE("Capture Volume", AK4535_PGA, 0, 127, 0),
+ SOC_SINGLE("Left Playback Volume", AK4535_LATT, 0, 127, 1),
+ SOC_SINGLE("Right Playback Volume", AK4535_RATT, 0, 127, 1),
+ SOC_SINGLE("AUX Bypass Volume", AK4535_VOL, 0, 15, 0),
+ SOC_SINGLE("Mic Sidetone Volume", AK4535_VOL, 4, 7, 0),
+};
+
+/* add non dapm controls */
+static int ak4535_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(ak4535_snd_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&ak4535_snd_controls[i], codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+/* Mono 1 Mixer */
+static const struct snd_kcontrol_new ak4535_mono1_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4535_SIG1, 4, 1, 0),
+ SOC_DAPM_SINGLE("Mono Playback Switch", AK4535_SIG1, 5, 1, 0),
+};
+
+/* Stereo Mixer */
+static const struct snd_kcontrol_new ak4535_stereo_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4535_SIG2, 4, 1, 0),
+ SOC_DAPM_SINGLE("Playback Switch", AK4535_SIG2, 7, 1, 0),
+ SOC_DAPM_SINGLE("Aux Bypass Switch", AK4535_SIG2, 5, 1, 0),
+};
+
+/* Input Mixer */
+static const struct snd_kcontrol_new ak4535_input_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Mic Capture Switch", AK4535_MIC, 2, 1, 0),
+ SOC_DAPM_SINGLE("Aux Capture Switch", AK4535_MIC, 5, 1, 0),
+};
+
+/* Input mux */
+static const struct snd_kcontrol_new ak4535_input_mux_control =
+ SOC_DAPM_ENUM("Input Select", ak4535_enum[4]);
+
+/* HP L switch */
+static const struct snd_kcontrol_new ak4535_hpl_control =
+ SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 1, 1, 1);
+
+/* HP R switch */
+static const struct snd_kcontrol_new ak4535_hpr_control =
+ SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 0, 1, 1);
+
+/* mono 2 switch */
+static const struct snd_kcontrol_new ak4535_mono2_control =
+ SOC_DAPM_SINGLE("Switch", AK4535_SIG1, 0, 1, 0);
+
+/* Line out switch */
+static const struct snd_kcontrol_new ak4535_line_control =
+ SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 6, 1, 0);
+
+/* ak4535 dapm widgets */
+static const struct snd_soc_dapm_widget ak4535_dapm_widgets[] = {
+ SND_SOC_DAPM_MIXER("Stereo Mixer", SND_SOC_NOPM, 0, 0,
+ &ak4535_stereo_mixer_controls[0],
+ ARRAY_SIZE(ak4535_stereo_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Mono1 Mixer", SND_SOC_NOPM, 0, 0,
+ &ak4535_mono1_mixer_controls[0],
+ ARRAY_SIZE(ak4535_mono1_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0,
+ &ak4535_input_mixer_controls[0],
+ ARRAY_SIZE(ak4535_input_mixer_controls)),
+ SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0,
+ &ak4535_input_mux_control),
+ SND_SOC_DAPM_DAC("DAC", "Playback", AK4535_PM2, 0, 0),
+ SND_SOC_DAPM_SWITCH("Mono 2 Enable", SND_SOC_NOPM, 0, 0,
+ &ak4535_mono2_control),
+ /* speaker powersave bit */
+ SND_SOC_DAPM_PGA("Speaker Enable", AK4535_MODE2, 0, 0, NULL, 0),
+ SND_SOC_DAPM_SWITCH("Line Out Enable", SND_SOC_NOPM, 0, 0,
+ &ak4535_line_control),
+ SND_SOC_DAPM_SWITCH("Left HP Enable", SND_SOC_NOPM, 0, 0,
+ &ak4535_hpl_control),
+ SND_SOC_DAPM_SWITCH("Right HP Enable", SND_SOC_NOPM, 0, 0,
+ &ak4535_hpr_control),
+ SND_SOC_DAPM_OUTPUT("LOUT"),
+ SND_SOC_DAPM_OUTPUT("HPL"),
+ SND_SOC_DAPM_OUTPUT("ROUT"),
+ SND_SOC_DAPM_OUTPUT("HPR"),
+ SND_SOC_DAPM_OUTPUT("SPP"),
+ SND_SOC_DAPM_OUTPUT("SPN"),
+ SND_SOC_DAPM_OUTPUT("MOUT1"),
+ SND_SOC_DAPM_OUTPUT("MOUT2"),
+ SND_SOC_DAPM_OUTPUT("MICOUT"),
+ SND_SOC_DAPM_ADC("ADC", "Capture", AK4535_PM1, 0, 0),
+ SND_SOC_DAPM_PGA("Spk Amp", AK4535_PM2, 3, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("HP R Amp", AK4535_PM2, 1, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("HP L Amp", AK4535_PM2, 2, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Mic", AK4535_PM1, 1, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Line Out", AK4535_PM1, 4, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Mono Out", AK4535_PM1, 3, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("AUX In", AK4535_PM1, 2, 0, NULL, 0),
+
+ SND_SOC_DAPM_MICBIAS("Mic Int Bias", AK4535_MIC, 3, 0),
+ SND_SOC_DAPM_MICBIAS("Mic Ext Bias", AK4535_MIC, 4, 0),
+ SND_SOC_DAPM_INPUT("MICIN"),
+ SND_SOC_DAPM_INPUT("MICEXT"),
+ SND_SOC_DAPM_INPUT("AUX"),
+ SND_SOC_DAPM_INPUT("MIN"),
+ SND_SOC_DAPM_INPUT("AIN"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ /*stereo mixer */
+ {"Stereo Mixer", "Playback Switch", "DAC"},
+ {"Stereo Mixer", "Mic Sidetone Switch", "Mic"},
+ {"Stereo Mixer", "Aux Bypass Switch", "AUX In"},
+
+ /* mono1 mixer */
+ {"Mono1 Mixer", "Mic Sidetone Switch", "Mic"},
+ {"Mono1 Mixer", "Mono Playback Switch", "DAC"},
+
+ /* Mic */
+ {"Mic", NULL, "AIN"},
+ {"Input Mux", "Internal", "Mic Int Bias"},
+ {"Input Mux", "External", "Mic Ext Bias"},
+ {"Mic Int Bias", NULL, "MICIN"},
+ {"Mic Ext Bias", NULL, "MICEXT"},
+ {"MICOUT", NULL, "Input Mux"},
+
+ /* line out */
+ {"LOUT", NULL, "Line Out Enable"},
+ {"ROUT", NULL, "Line Out Enable"},
+ {"Line Out Enable", "Switch", "Line Out"},
+ {"Line Out", NULL, "Stereo Mixer"},
+
+ /* mono1 out */
+ {"MOUT1", NULL, "Mono Out"},
+ {"Mono Out", NULL, "Mono1 Mixer"},
+
+ /* left HP */
+ {"HPL", NULL, "Left HP Enable"},
+ {"Left HP Enable", "Switch", "HP L Amp"},
+ {"HP L Amp", NULL, "Stereo Mixer"},
+
+ /* right HP */
+ {"HPR", NULL, "Right HP Enable"},
+ {"Right HP Enable", "Switch", "HP R Amp"},
+ {"HP R Amp", NULL, "Stereo Mixer"},
+
+ /* speaker */
+ {"SPP", NULL, "Speaker Enable"},
+ {"SPN", NULL, "Speaker Enable"},
+ {"Speaker Enable", "Switch", "Spk Amp"},
+ {"Spk Amp", NULL, "MIN"},
+
+ /* mono 2 */
+ {"MOUT2", NULL, "Mono 2 Enable"},
+ {"Mono 2 Enable", "Switch", "Stereo Mixer"},
+
+ /* Aux In */
+ {"Aux In", NULL, "AUX"},
+
+ /* ADC */
+ {"ADC", NULL, "Input Mixer"},
+ {"Input Mixer", "Mic Capture Switch", "Mic"},
+ {"Input Mixer", "Aux Capture Switch", "Aux In"},
+};
+
+static int ak4535_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, ak4535_dapm_widgets,
+ ARRAY_SIZE(ak4535_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+static int ak4535_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 ak4535_priv *ak4535 = codec->private_data;
+
+ ak4535->sysclk = freq;
+ return 0;
+}
+
+static int ak4535_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;
+ struct ak4535_priv *ak4535 = codec->private_data;
+ u8 mode2 = ak4535_read_reg_cache(codec, AK4535_MODE2) & ~(0x3 << 5);
+ int rate = params_rate(params), fs = 256;
+
+ if (rate)
+ fs = ak4535->sysclk / rate;
+
+ /* set fs */
+ switch (fs) {
+ case 1024:
+ mode2 |= (0x2 << 5);
+ break;
+ case 512:
+ mode2 |= (0x1 << 5);
+ break;
+ case 256:
+ break;
+ }
+
+ /* set rate */
+ ak4535_write(codec, AK4535_MODE2, mode2);
+ return 0;
+}
+
+static int ak4535_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u8 mode1 = 0;
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ mode1 = 0x0002;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ mode1 = 0x0001;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* use 32 fs for BCLK to save power */
+ mode1 |= 0x4;
+
+ ak4535_write(codec, AK4535_MODE1, mode1);
+ return 0;
+}
+
+static int ak4535_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC) & 0xffdf;
+ if (!mute)
+ ak4535_write(codec, AK4535_DAC, mute_reg);
+ else
+ ak4535_write(codec, AK4535_DAC, mute_reg | 0x20);
+ return 0;
+}
+
+static int ak4535_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u16 i;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ ak4535_mute(codec->dai, 0);
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ ak4535_mute(codec->dai, 1);
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ i = ak4535_read_reg_cache(codec, AK4535_PM1);
+ ak4535_write(codec, AK4535_PM1, i | 0x80);
+ i = ak4535_read_reg_cache(codec, AK4535_PM2);
+ ak4535_write(codec, AK4535_PM2, i & (~0x80));
+ break;
+ case SND_SOC_BIAS_OFF:
+ i = ak4535_read_reg_cache(codec, AK4535_PM1);
+ ak4535_write(codec, AK4535_PM1, i & (~0x80));
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+#define AK4535_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
+
+struct snd_soc_dai ak4535_dai = {
+ .name = "AK4535",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AK4535_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AK4535_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
+ .hw_params = ak4535_hw_params,
+ },
+ .dai_ops = {
+ .set_fmt = ak4535_set_dai_fmt,
+ .digital_mute = ak4535_mute,
+ .set_sysclk = ak4535_set_dai_sysclk,
+ },
+};
+EXPORT_SYMBOL_GPL(ak4535_dai);
+
+static int ak4535_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;
+
+ ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int ak4535_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ ak4535_sync(codec);
+ ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ ak4535_set_bias_level(codec, codec->suspend_bias_level);
+ return 0;
+}
+
+/*
+ * initialise the AK4535 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int ak4535_init(struct snd_soc_device *socdev)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret = 0;
+
+ codec->name = "AK4535";
+ codec->owner = THIS_MODULE;
+ codec->read = ak4535_read_reg_cache;
+ codec->write = ak4535_write;
+ codec->set_bias_level = ak4535_set_bias_level;
+ codec->dai = &ak4535_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = ARRAY_SIZE(ak4535_reg);
+ codec->reg_cache = kmemdup(ak4535_reg, sizeof(ak4535_reg), GFP_KERNEL);
+
+ if (codec->reg_cache == NULL)
+ return -ENOMEM;
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ printk(KERN_ERR "ak4535: failed to create pcms\n");
+ goto pcm_err;
+ }
+
+ /* power on device */
+ ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ ak4535_add_controls(codec);
+ ak4535_add_widgets(codec);
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "ak4535: 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 *ak4535_socdev;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+
+#define I2C_DRIVERID_AK4535 0xfefe /* liam - need a proper id */
+
+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
+
+/* Magic definition of all other variables and things */
+I2C_CLIENT_INSMOD;
+
+static struct i2c_driver ak4535_i2c_driver;
+static struct i2c_client client_template;
+
+/* If the i2c layer weren't so broken, we could pass this kind of data
+ around */
+static int ak4535_codec_probe(struct i2c_adapter *adap, int addr, int kind)
+{
+ struct snd_soc_device *socdev = ak4535_socdev;
+ struct ak4535_setup_data *setup = socdev->codec_data;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct i2c_client *i2c;
+ int ret;
+
+ if (addr != setup->i2c_address)
+ return -ENODEV;
+
+ client_template.adapter = adap;
+ client_template.addr = addr;
+
+ i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL);
+ if (i2c == NULL) {
+ kfree(codec);
+ return -ENOMEM;
+ }
+ i2c_set_clientdata(i2c, codec);
+ codec->control_data = i2c;
+
+ ret = i2c_attach_client(i2c);
+ if (ret < 0) {
+ printk(KERN_ERR "failed to attach codec at addr %x\n", addr);
+ goto err;
+ }
+
+ ret = ak4535_init(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "failed to initialise AK4535\n");
+ goto err;
+ }
+ return ret;
+
+err:
+ kfree(codec);
+ kfree(i2c);
+ return ret;
+}
+
+static int ak4535_i2c_detach(struct i2c_client *client)
+{
+ struct snd_soc_codec *codec = i2c_get_clientdata(client);
+ i2c_detach_client(client);
+ kfree(codec->reg_cache);
+ kfree(client);
+ return 0;
+}
+
+static int ak4535_i2c_attach(struct i2c_adapter *adap)
+{
+ return i2c_probe(adap, &addr_data, ak4535_codec_probe);
+}
+
+/* corgi i2c codec control layer */
+static struct i2c_driver ak4535_i2c_driver = {
+ .driver = {
+ .name = "AK4535 I2C Codec",
+ .owner = THIS_MODULE,
+ },
+ .id = I2C_DRIVERID_AK4535,
+ .attach_adapter = ak4535_i2c_attach,
+ .detach_client = ak4535_i2c_detach,
+ .command = NULL,
+};
+
+static struct i2c_client client_template = {
+ .name = "AK4535",
+ .driver = &ak4535_i2c_driver,
+};
+#endif
+
+static int ak4535_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct ak4535_setup_data *setup;
+ struct snd_soc_codec *codec;
+ struct ak4535_priv *ak4535;
+ int ret = 0;
+
+ printk(KERN_INFO "AK4535 Audio Codec %s", AK4535_VERSION);
+
+ setup = socdev->codec_data;
+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (codec == NULL)
+ return -ENOMEM;
+
+ ak4535 = kzalloc(sizeof(struct ak4535_priv), GFP_KERNEL);
+ if (ak4535 == NULL) {
+ kfree(codec);
+ return -ENOMEM;
+ }
+
+ codec->private_data = ak4535;
+ socdev->codec = codec;
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ ak4535_socdev = socdev;
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ if (setup->i2c_address) {
+ normal_i2c[0] = setup->i2c_address;
+ codec->hw_write = (hw_write_t)i2c_master_send;
+ codec->hw_read = (hw_read_t)i2c_master_recv;
+ ret = i2c_add_driver(&ak4535_i2c_driver);
+ if (ret != 0)
+ printk(KERN_ERR "can't add i2c driver");
+ }
+#else
+ /* Add other interfaces here */
+#endif
+ return ret;
+}
+
+/* power down chip */
+static int ak4535_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)
+ ak4535_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(&ak4535_i2c_driver);
+#endif
+ kfree(codec->private_data);
+ kfree(codec);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_ak4535 = {
+ .probe = ak4535_probe,
+ .remove = ak4535_remove,
+ .suspend = ak4535_suspend,
+ .resume = ak4535_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_ak4535);
+
+MODULE_DESCRIPTION("Soc AK4535 driver");
+MODULE_AUTHOR("Richard Purdie");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ak4535.h b/sound/soc/codecs/ak4535.h
new file mode 100644
index 00000000000..e9fe30e2c05
--- /dev/null
+++ b/sound/soc/codecs/ak4535.h
@@ -0,0 +1,46 @@
+/*
+ * ak4535.h -- AK4535 Soc Audio driver
+ *
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Author: Richard Purdie <richard@openedhand.com>
+ *
+ * Based on wm8753.h
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _AK4535_H
+#define _AK4535_H
+
+/* AK4535 register space */
+
+#define AK4535_PM1 0x0
+#define AK4535_PM2 0x1
+#define AK4535_SIG1 0x2
+#define AK4535_SIG2 0x3
+#define AK4535_MODE1 0x4
+#define AK4535_MODE2 0x5
+#define AK4535_DAC 0x6
+#define AK4535_MIC 0x7
+#define AK4535_TIMER 0x8
+#define AK4535_ALC1 0x9
+#define AK4535_ALC2 0xa
+#define AK4535_PGA 0xb
+#define AK4535_LATT 0xc
+#define AK4535_RATT 0xd
+#define AK4535_VOL 0xe
+#define AK4535_STATUS 0xf
+
+#define AK4535_CACHEREGNUM 0x10
+
+struct ak4535_setup_data {
+ unsigned short i2c_address;
+};
+
+extern struct snd_soc_dai ak4535_dai;
+extern struct snd_soc_codec_device soc_codec_dev_ak4535;
+
+#endif
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index e73fcfd9f5c..9deb8c74fdf 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -201,7 +201,7 @@ static struct {
* driver what the input settings can be. This would need to be implemented
* for stand-alone mode to work.
*/
-static int cs4270_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
+static int cs4270_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;
@@ -251,7 +251,7 @@ static int cs4270_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
* data for playback only, but ASoC currently does not support different
* formats for playback vs. record.
*/
-static int cs4270_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int format)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -471,7 +471,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
* board does not have the MUTEA or MUTEB pins connected to such circuitry,
* then this function will do nothing.
*/
-static int cs4270_mute(struct snd_soc_codec_dai *dai, int mute)
+static int cs4270_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
int reg6;
@@ -667,7 +667,7 @@ error:
#endif /* USE_I2C*/
-struct snd_soc_codec_dai cs4270_dai = {
+struct snd_soc_dai cs4270_dai = {
.name = "CS4270",
.playback = {
.stream_name = "Playback",
diff --git a/sound/soc/codecs/cs4270.h b/sound/soc/codecs/cs4270.h
index 0ced49b7804..adc6cd9667d 100644
--- a/sound/soc/codecs/cs4270.h
+++ b/sound/soc/codecs/cs4270.h
@@ -16,7 +16,7 @@
* The ASoC codec DAI structure for the CS4270. Assign this structure to
* the .codec_dai field of your machine driver's snd_soc_dai_link structure.
*/
-extern struct snd_soc_codec_dai cs4270_dai;
+extern struct snd_soc_dai cs4270_dai;
/*
* The ASoC codec device structure for the CS4270. Assign this structure
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 09b1661b8a3..b1dce5f459d 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -29,7 +29,7 @@
* ---------------------------------------
*
* Hence the machine layer should disable unsupported inputs/outputs by
- * snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0), etc.
+ * snd_soc_dapm_disable_pin(codec, "MONO_LOUT"), etc.
*/
#include <linux/module.h>
@@ -49,7 +49,7 @@
#include "tlv320aic3x.h"
#define AUDIO_NAME "aic3x"
-#define AIC3X_VERSION "0.1"
+#define AIC3X_VERSION "0.2"
/* codec private data */
struct aic3x_priv {
@@ -138,6 +138,20 @@ static int aic3x_write(struct snd_soc_codec *codec, unsigned int reg,
return -EIO;
}
+/*
+ * read from the aic3x register space
+ */
+static int aic3x_read(struct snd_soc_codec *codec, unsigned int reg,
+ u8 *value)
+{
+ *value = reg & 0xff;
+ if (codec->hw_read(codec->control_data, value, 1) != 1)
+ return -EIO;
+
+ aic3x_write_reg_cache(codec, reg, *value);
+ return 0;
+}
+
#define SOC_DAPM_SINGLE_AIC3X(xname, reg, shift, mask, invert) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.info = snd_soc_info_volsw, \
@@ -192,7 +206,7 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol,
}
if (found)
- snd_soc_dapm_sync_endpoints(widget->codec);
+ snd_soc_dapm_sync(widget->codec);
}
ret = snd_soc_update_bits(widget->codec, reg, val_mask, val);
@@ -209,6 +223,8 @@ static const char *aic3x_right_hpcom_mux[] =
{ "differential of HPROUT", "constant VCM", "single-ended",
"differential of HPLCOM", "external feedback" };
static const char *aic3x_linein_mode_mux[] = { "single-ended", "differential" };
+static const char *aic3x_adc_hpf[] =
+ { "Disabled", "0.0045xFs", "0.0125xFs", "0.025xFs" };
#define LDAC_ENUM 0
#define RDAC_ENUM 1
@@ -218,6 +234,7 @@ static const char *aic3x_linein_mode_mux[] = { "single-ended", "differential" };
#define LINE1R_ENUM 5
#define LINE2L_ENUM 6
#define LINE2R_ENUM 7
+#define ADC_HPF_ENUM 8
static const struct soc_enum aic3x_enum[] = {
SOC_ENUM_SINGLE(DAC_LINE_MUX, 6, 3, aic3x_left_dac_mux),
@@ -228,6 +245,7 @@ static const struct soc_enum aic3x_enum[] = {
SOC_ENUM_SINGLE(LINE1R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux),
SOC_ENUM_SINGLE(LINE2L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux),
SOC_ENUM_SINGLE(LINE2R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux),
+ SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf),
};
static const struct snd_kcontrol_new aic3x_snd_controls[] = {
@@ -278,6 +296,8 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
/* Input */
SOC_DOUBLE_R("PGA Capture Volume", LADC_VOL, RADC_VOL, 0, 0x7f, 0),
SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1),
+
+ SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]),
};
/* add non dapm controls */
@@ -441,11 +461,34 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
SND_SOC_DAPM_MUX("Right Line2R Mux", SND_SOC_NOPM, 0, 0,
&aic3x_right_line2_mux_controls),
+ /*
+ * Not a real mic bias widget but similar function. This is for dynamic
+ * control of GPIO1 digital mic modulator clock output function when
+ * using digital mic.
+ */
+ SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "GPIO1 dmic modclk",
+ AIC3X_GPIO1_REG, 4, 0xf,
+ AIC3X_GPIO1_FUNC_DIGITAL_MIC_MODCLK,
+ AIC3X_GPIO1_FUNC_DISABLED),
+
+ /*
+ * Also similar function like mic bias. Selects digital mic with
+ * configurable oversampling rate instead of ADC converter.
+ */
+ SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 128",
+ AIC3X_ASD_INTF_CTRLA, 0, 3, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 64",
+ AIC3X_ASD_INTF_CTRLA, 0, 3, 2, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 32",
+ AIC3X_ASD_INTF_CTRLA, 0, 3, 3, 0),
+
/* Mic Bias */
- SND_SOC_DAPM_MICBIAS("Mic Bias 2V", MICBIAS_CTRL, 6, 0),
- SND_SOC_DAPM_MICBIAS("Mic Bias 2.5V", MICBIAS_CTRL, 7, 0),
- SND_SOC_DAPM_MICBIAS("Mic Bias AVDD", MICBIAS_CTRL, 6, 0),
- SND_SOC_DAPM_MICBIAS("Mic Bias AVDD", MICBIAS_CTRL, 7, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias 2V",
+ MICBIAS_CTRL, 6, 3, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias 2.5V",
+ MICBIAS_CTRL, 6, 3, 2, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias AVDD",
+ MICBIAS_CTRL, 6, 3, 3, 0),
/* Left PGA to Left Output bypass */
SND_SOC_DAPM_MIXER("Left PGA Bypass Mixer", SND_SOC_NOPM, 0, 0,
@@ -483,7 +526,7 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
SND_SOC_DAPM_INPUT("LINE2R"),
};
-static const char *intercon[][3] = {
+static const struct snd_soc_dapm_route intercon[] = {
/* Left Output */
{"Left DAC Mux", "DAC_L1", "Left DAC"},
{"Left DAC Mux", "DAC_L2", "Left DAC"},
@@ -554,6 +597,7 @@ static const char *intercon[][3] = {
{"Left PGA Mixer", "Mic3L Switch", "MIC3L"},
{"Left ADC", NULL, "Left PGA Mixer"},
+ {"Left ADC", NULL, "GPIO1 dmic modclk"},
/* Right Input */
{"Right Line1R Mux", "single-ended", "LINE1R"},
@@ -567,6 +611,7 @@ static const char *intercon[][3] = {
{"Right PGA Mixer", "Mic3R Switch", "MIC3R"},
{"Right ADC", NULL, "Right PGA Mixer"},
+ {"Right ADC", NULL, "GPIO1 dmic modclk"},
/* Left PGA Bypass */
{"Left PGA Bypass Mixer", "Line Switch", "Left PGA Mixer"},
@@ -628,101 +673,27 @@ static const char *intercon[][3] = {
{"Mono Out", NULL, "Right Line2 Bypass Mixer"},
{"Right HP Out", NULL, "Right Line2 Bypass Mixer"},
- /* terminator */
- {NULL, NULL, NULL},
+ /*
+ * Logical path between digital mic enable and GPIO1 modulator clock
+ * output function
+ */
+ {"GPIO1 dmic modclk", NULL, "DMic Rate 128"},
+ {"GPIO1 dmic modclk", NULL, "DMic Rate 64"},
+ {"GPIO1 dmic modclk", NULL, "DMic Rate 32"},
};
static int aic3x_add_widgets(struct snd_soc_codec *codec)
{
- int i;
-
- for (i = 0; i < ARRAY_SIZE(aic3x_dapm_widgets); i++)
- snd_soc_dapm_new_control(codec, &aic3x_dapm_widgets[i]);
+ snd_soc_dapm_new_controls(codec, aic3x_dapm_widgets,
+ ARRAY_SIZE(aic3x_dapm_widgets));
/* set up audio path interconnects */
- for (i = 0; intercon[i][0] != NULL; i++)
- snd_soc_dapm_connect_input(codec, intercon[i][0],
- intercon[i][1], intercon[i][2]);
+ snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
snd_soc_dapm_new_widgets(codec);
return 0;
}
-struct aic3x_rate_divs {
- u32 mclk;
- u32 rate;
- u32 fsref_reg;
- u8 sr_reg:4;
- u8 pllj_reg;
- u16 plld_reg;
-};
-
-/* AIC3X codec mclk clock divider coefficients */
-static const struct aic3x_rate_divs aic3x_divs[] = {
- /* 8k */
- {12000000, 8000, 48000, 0xa, 16, 3840},
- {19200000, 8000, 48000, 0xa, 10, 2400},
- {22579200, 8000, 48000, 0xa, 8, 7075},
- {33868800, 8000, 48000, 0xa, 5, 8049},
- /* 11.025k */
- {12000000, 11025, 44100, 0x6, 15, 528},
- {19200000, 11025, 44100, 0x6, 9, 4080},
- {22579200, 11025, 44100, 0x6, 8, 0},
- {33868800, 11025, 44100, 0x6, 5, 3333},
- /* 16k */
- {12000000, 16000, 48000, 0x4, 16, 3840},
- {19200000, 16000, 48000, 0x4, 10, 2400},
- {22579200, 16000, 48000, 0x4, 8, 7075},
- {33868800, 16000, 48000, 0x4, 5, 8049},
- /* 22.05k */
- {12000000, 22050, 44100, 0x2, 15, 528},
- {19200000, 22050, 44100, 0x2, 9, 4080},
- {22579200, 22050, 44100, 0x2, 8, 0},
- {33868800, 22050, 44100, 0x2, 5, 3333},
- /* 32k */
- {12000000, 32000, 48000, 0x1, 16, 3840},
- {19200000, 32000, 48000, 0x1, 10, 2400},
- {22579200, 32000, 48000, 0x1, 8, 7075},
- {33868800, 32000, 48000, 0x1, 5, 8049},
- /* 44.1k */
- {12000000, 44100, 44100, 0x0, 15, 528},
- {19200000, 44100, 44100, 0x0, 9, 4080},
- {22579200, 44100, 44100, 0x0, 8, 0},
- {33868800, 44100, 44100, 0x0, 5, 3333},
- /* 48k */
- {12000000, 48000, 48000, 0x0, 16, 3840},
- {19200000, 48000, 48000, 0x0, 10, 2400},
- {22579200, 48000, 48000, 0x0, 8, 7075},
- {33868800, 48000, 48000, 0x0, 5, 8049},
- /* 64k */
- {12000000, 64000, 96000, 0x1, 16, 3840},
- {19200000, 64000, 96000, 0x1, 10, 2400},
- {22579200, 64000, 96000, 0x1, 8, 7075},
- {33868800, 64000, 96000, 0x1, 5, 8049},
- /* 88.2k */
- {12000000, 88200, 88200, 0x0, 15, 528},
- {19200000, 88200, 88200, 0x0, 9, 4080},
- {22579200, 88200, 88200, 0x0, 8, 0},
- {33868800, 88200, 88200, 0x0, 5, 3333},
- /* 96k */
- {12000000, 96000, 96000, 0x0, 16, 3840},
- {19200000, 96000, 96000, 0x0, 10, 2400},
- {22579200, 96000, 96000, 0x0, 8, 7075},
- {33868800, 96000, 96000, 0x0, 5, 8049},
-};
-
-static inline int aic3x_get_divs(int mclk, int rate)
-{
- int i;
-
- for (i = 0; i < ARRAY_SIZE(aic3x_divs); i++) {
- if (aic3x_divs[i].rate == rate && aic3x_divs[i].mclk == mclk)
- return i;
- }
-
- return 0;
-}
-
static int aic3x_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
@@ -730,49 +701,107 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->codec;
struct aic3x_priv *aic3x = codec->private_data;
- int i;
- u8 data, pll_p, pll_r, pll_j;
- u16 pll_d;
-
- i = aic3x_get_divs(aic3x->sysclk, params_rate(params));
+ int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0;
+ u8 data, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1;
+ u16 pll_d = 1;
- /* Route Left DAC to left channel input and
- * right DAC to right channel input */
- data = (LDAC2LCH | RDAC2RCH);
- switch (aic3x_divs[i].fsref_reg) {
- case 44100:
- data |= FSREF_44100;
+ /* select data word length */
+ data =
+ aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLB) & (~(0x3 << 4));
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
break;
- case 48000:
- data |= FSREF_48000;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ data |= (0x01 << 4);
break;
- case 88200:
- data |= FSREF_44100 | DUAL_RATE_MODE;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ data |= (0x02 << 4);
break;
- case 96000:
- data |= FSREF_48000 | DUAL_RATE_MODE;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ data |= (0x03 << 4);
break;
}
+ aic3x_write(codec, AIC3X_ASD_INTF_CTRLB, data);
+
+ /* Fsref can be 44100 or 48000 */
+ fsref = (params_rate(params) % 11025 == 0) ? 44100 : 48000;
+
+ /* Try to find a value for Q which allows us to bypass the PLL and
+ * generate CODEC_CLK directly. */
+ for (pll_q = 2; pll_q < 18; pll_q++)
+ if (aic3x->sysclk / (128 * pll_q) == fsref) {
+ bypass_pll = 1;
+ break;
+ }
+
+ if (bypass_pll) {
+ pll_q &= 0xf;
+ aic3x_write(codec, AIC3X_PLL_PROGA_REG, pll_q << PLLQ_SHIFT);
+ aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_CLKDIV);
+ } else
+ aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_PLLDIV);
+
+ /* Route Left DAC to left channel input and
+ * right DAC to right channel input */
+ data = (LDAC2LCH | RDAC2RCH);
+ data |= (fsref == 44100) ? FSREF_44100 : FSREF_48000;
+ if (params_rate(params) >= 64000)
+ data |= DUAL_RATE_MODE;
aic3x_write(codec, AIC3X_CODEC_DATAPATH_REG, data);
/* codec sample rate select */
- data = aic3x_divs[i].sr_reg;
+ data = (fsref * 20) / params_rate(params);
+ if (params_rate(params) < 64000)
+ data /= 2;
+ data /= 5;
+ data -= 2;
data |= (data << 4);
aic3x_write(codec, AIC3X_SAMPLE_RATE_SEL_REG, data);
- /* Use PLL for generation Fsref by equation:
- * Fsref = (MCLK * K * R)/(2048 * P);
- * Fix P = 2 and R = 1 and calculate K, if
- * K = J.D, i.e. J - an interger portion of K and D is the fractional
- * one with 4 digits of precision;
- * Example:
- * For MCLK = 22.5792 MHz and Fsref = 48kHz:
- * Select P = 2, R= 1, K = 8.7074, which results in J = 8, D = 7074
+ if (bypass_pll)
+ return 0;
+
+ /* Use PLL
+ * find an apropriate setup for j, d, r and p by iterating over
+ * p and r - j and d are calculated for each fraction.
+ * Up to 128 values are probed, the closest one wins the game.
+ * The sysclk is divided by 1000 to prevent integer overflows.
*/
- pll_p = 2;
- pll_r = 1;
- pll_j = aic3x_divs[i].pllj_reg;
- pll_d = aic3x_divs[i].plld_reg;
+ codec_clk = (2048 * fsref) / (aic3x->sysclk / 1000);
+
+ for (r = 1; r <= 16; r++)
+ for (p = 1; p <= 8; p++) {
+ int clk, tmp = (codec_clk * pll_r * 10) / pll_p;
+ u8 j = tmp / 10000;
+ u16 d = tmp % 10000;
+
+ if (j > 63)
+ continue;
+
+ if (d != 0 && aic3x->sysclk < 10000000)
+ continue;
+
+ /* This is actually 1000 * ((j + (d/10000)) * r) / p
+ * The term had to be converted to get rid of the
+ * division by 10000 */
+ clk = ((10000 * j * r) + (d * r)) / (10 * p);
+
+ /* check whether this values get closer than the best
+ * ones we had before */
+ if (abs(codec_clk - clk) < abs(codec_clk - last_clk)) {
+ pll_j = j; pll_d = d; pll_r = r; pll_p = p;
+ last_clk = clk;
+ }
+
+ /* Early exit for exact matches */
+ if (clk == codec_clk)
+ break;
+ }
+
+ if (last_clk == 0) {
+ printk(KERN_ERR "%s(): unable to setup PLL\n", __func__);
+ return -EINVAL;
+ }
data = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG);
aic3x_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT));
@@ -782,28 +811,10 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
aic3x_write(codec, AIC3X_PLL_PROGD_REG,
(pll_d & 0x3F) << PLLD_LSB_SHIFT);
- /* select data word length */
- data =
- aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLB) & (~(0x3 << 4));
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
- break;
- case SNDRV_PCM_FORMAT_S20_3LE:
- data |= (0x01 << 4);
- break;
- case SNDRV_PCM_FORMAT_S24_LE:
- data |= (0x02 << 4);
- break;
- case SNDRV_PCM_FORMAT_S32_LE:
- data |= (0x03 << 4);
- break;
- }
- aic3x_write(codec, AIC3X_ASD_INTF_CTRLB, data);
-
return 0;
}
-static int aic3x_mute(struct snd_soc_codec_dai *dai, int mute)
+static int aic3x_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
u8 ldac_reg = aic3x_read_reg_cache(codec, LDAC_VOL) & ~MUTE_ON;
@@ -820,31 +831,25 @@ static int aic3x_mute(struct snd_soc_codec_dai *dai, int mute)
return 0;
}
-static int aic3x_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
+static int aic3x_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 aic3x_priv *aic3x = codec->private_data;
- switch (freq) {
- case 12000000:
- case 19200000:
- case 22579200:
- case 33868800:
- aic3x->sysclk = freq;
- return 0;
- }
-
- return -EINVAL;
+ aic3x->sysclk = freq;
+ return 0;
}
-static int aic3x_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
struct aic3x_priv *aic3x = codec->private_data;
- u8 iface_areg = 0;
- u8 iface_breg = 0;
+ u8 iface_areg, iface_breg;
+
+ iface_areg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLA) & 0x3f;
+ iface_breg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLB) & 0x3f;
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
@@ -883,13 +888,14 @@ static int aic3x_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
return 0;
}
-static int aic3x_dapm_event(struct snd_soc_codec *codec, int event)
+static int aic3x_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
{
struct aic3x_priv *aic3x = codec->private_data;
u8 reg;
- switch (event) {
- case SNDRV_CTL_POWER_D0:
+ switch (level) {
+ case SND_SOC_BIAS_ON:
/* all power is driven by DAPM system */
if (aic3x->master) {
/* enable pll */
@@ -898,10 +904,9 @@ static int aic3x_dapm_event(struct snd_soc_codec *codec, int event)
reg | PLL_ENABLE);
}
break;
- case SNDRV_CTL_POWER_D1:
- case SNDRV_CTL_POWER_D2:
+ case SND_SOC_BIAS_PREPARE:
break;
- case SNDRV_CTL_POWER_D3hot:
+ case SND_SOC_BIAS_STANDBY:
/*
* all power is driven by DAPM system,
* so output power is safe if bypass was set
@@ -913,7 +918,7 @@ static int aic3x_dapm_event(struct snd_soc_codec *codec, int event)
reg & ~PLL_ENABLE);
}
break;
- case SNDRV_CTL_POWER_D3cold:
+ case SND_SOC_BIAS_OFF:
/* force all power off */
reg = aic3x_read_reg_cache(codec, LINE1L_2_LADC_CTRL);
aic3x_write(codec, LINE1L_2_LADC_CTRL, reg & ~LADC_PWR_ON);
@@ -949,16 +954,43 @@ static int aic3x_dapm_event(struct snd_soc_codec *codec, int event)
}
break;
}
- codec->dapm_state = event;
+ codec->bias_level = level;
return 0;
}
+void aic3x_set_gpio(struct snd_soc_codec *codec, int gpio, int state)
+{
+ u8 reg = gpio ? AIC3X_GPIO2_REG : AIC3X_GPIO1_REG;
+ u8 bit = gpio ? 3: 0;
+ u8 val = aic3x_read_reg_cache(codec, reg) & ~(1 << bit);
+ aic3x_write(codec, reg, val | (!!state << bit));
+}
+EXPORT_SYMBOL_GPL(aic3x_set_gpio);
+
+int aic3x_get_gpio(struct snd_soc_codec *codec, int gpio)
+{
+ u8 reg = gpio ? AIC3X_GPIO2_REG : AIC3X_GPIO1_REG;
+ u8 val, bit = gpio ? 2: 1;
+
+ aic3x_read(codec, reg, &val);
+ return (val >> bit) & 1;
+}
+EXPORT_SYMBOL_GPL(aic3x_get_gpio);
+
+int aic3x_headset_detected(struct snd_soc_codec *codec)
+{
+ u8 val;
+ aic3x_read(codec, AIC3X_RT_IRQ_FLAGS_REG, &val);
+ return (val >> 2) & 1;
+}
+EXPORT_SYMBOL_GPL(aic3x_headset_detected);
+
#define AIC3X_RATES SNDRV_PCM_RATE_8000_96000
#define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
-struct snd_soc_codec_dai aic3x_dai = {
+struct snd_soc_dai aic3x_dai = {
.name = "aic3x",
.playback = {
.stream_name = "Playback",
@@ -988,7 +1020,7 @@ static int aic3x_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;
- aic3x_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+ aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
@@ -1008,7 +1040,7 @@ static int aic3x_resume(struct platform_device *pdev)
codec->hw_write(codec->control_data, data, 2);
}
- aic3x_dapm_event(codec, codec->suspend_dapm_state);
+ aic3x_set_bias_level(codec, codec->suspend_bias_level);
return 0;
}
@@ -1020,16 +1052,17 @@ static int aic3x_resume(struct platform_device *pdev)
static int aic3x_init(struct snd_soc_device *socdev)
{
struct snd_soc_codec *codec = socdev->codec;
+ struct aic3x_setup_data *setup = socdev->codec_data;
int reg, ret = 0;
codec->name = "aic3x";
codec->owner = THIS_MODULE;
codec->read = aic3x_read_reg_cache;
codec->write = aic3x_write;
- codec->dapm_event = aic3x_dapm_event;
+ codec->set_bias_level = aic3x_set_bias_level;
codec->dai = &aic3x_dai;
codec->num_dai = 1;
- codec->reg_cache_size = sizeof(aic3x_reg);
+ codec->reg_cache_size = ARRAY_SIZE(aic3x_reg);
codec->reg_cache = kmemdup(aic3x_reg, sizeof(aic3x_reg), GFP_KERNEL);
if (codec->reg_cache == NULL)
return -ENOMEM;
@@ -1108,7 +1141,11 @@ static int aic3x_init(struct snd_soc_device *socdev)
aic3x_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL);
/* off, with power on */
- aic3x_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+ aic3x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ /* setup GPIO functions */
+ aic3x_write(codec, AIC3X_GPIO1_REG, (setup->gpio_func[0] & 0xf) << 4);
+ aic3x_write(codec, AIC3X_GPIO2_REG, (setup->gpio_func[1] & 0xf) << 4);
aic3x_add_controls(codec);
aic3x_add_widgets(codec);
@@ -1217,6 +1254,12 @@ static struct i2c_client client_template = {
.name = "AIC3X",
.driver = &aic3x_i2c_driver,
};
+
+static int aic3x_i2c_read(struct i2c_client *client, u8 *value, int len)
+{
+ value[0] = i2c_smbus_read_byte_data(client, value[0]);
+ return (len == 1);
+}
#endif
static int aic3x_probe(struct platform_device *pdev)
@@ -1251,6 +1294,7 @@ static int aic3x_probe(struct platform_device *pdev)
if (setup->i2c_address) {
normal_i2c[0] = setup->i2c_address;
codec->hw_write = (hw_write_t) i2c_master_send;
+ codec->hw_read = (hw_read_t) aic3x_i2c_read;
ret = i2c_add_driver(&aic3x_i2c_driver);
if (ret != 0)
printk(KERN_ERR "can't add i2c driver");
@@ -1268,7 +1312,7 @@ static int aic3x_remove(struct platform_device *pdev)
/* power down chip */
if (codec->control_data)
- aic3x_dapm_event(codec, SNDRV_CTL_POWER_D3);
+ aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF);
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h
index d0cdeeb629d..d76c079b86e 100644
--- a/sound/soc/codecs/tlv320aic3x.h
+++ b/sound/soc/codecs/tlv320aic3x.h
@@ -37,6 +37,8 @@
#define AIC3X_ASD_INTF_CTRLB 9
/* Audio overflow status and PLL R value programming register */
#define AIC3X_OVRF_STATUS_AND_PLLR_REG 11
+/* Audio codec digital filter control register */
+#define AIC3X_CODEC_DFILT_CTRL 12
/* ADC PGA Gain control registers */
#define LADC_VOL 15
@@ -108,6 +110,13 @@
#define DACR1_2_RLOPM_VOL 92
#define LLOPM_CTRL 86
#define RLOPM_CTRL 93
+/* GPIO/IRQ registers */
+#define AIC3X_STICKY_IRQ_FLAGS_REG 96
+#define AIC3X_RT_IRQ_FLAGS_REG 97
+#define AIC3X_GPIO1_REG 98
+#define AIC3X_GPIO2_REG 99
+#define AIC3X_GPIOA_REG 100
+#define AIC3X_GPIOB_REG 101
/* Clock generation control register */
#define AIC3X_CLKGEN_CTRL_REG 102
@@ -128,12 +137,15 @@
/* PLL registers bitfields */
#define PLLP_SHIFT 0
+#define PLLQ_SHIFT 3
#define PLLR_SHIFT 0
#define PLLJ_SHIFT 2
#define PLLD_MSB_SHIFT 0
#define PLLD_LSB_SHIFT 2
/* Clock generation register bits */
+#define CODEC_CLKIN_PLLDIV 0
+#define CODEC_CLKIN_CLKDIV 1
#define PLL_CLKIN_SHIFT 4
#define MCLK_SOURCE 0x0
#define PLL_CLKDIV_SHIFT 0
@@ -171,11 +183,52 @@
/* Default input volume */
#define DEFAULT_GAIN 0x20
+/* GPIO API */
+enum {
+ AIC3X_GPIO1_FUNC_DISABLED = 0,
+ AIC3X_GPIO1_FUNC_AUDIO_WORDCLK_ADC = 1,
+ AIC3X_GPIO1_FUNC_CLOCK_MUX = 2,
+ AIC3X_GPIO1_FUNC_CLOCK_MUX_DIV2 = 3,
+ AIC3X_GPIO1_FUNC_CLOCK_MUX_DIV4 = 4,
+ AIC3X_GPIO1_FUNC_CLOCK_MUX_DIV8 = 5,
+ AIC3X_GPIO1_FUNC_SHORT_CIRCUIT_IRQ = 6,
+ AIC3X_GPIO1_FUNC_AGC_NOISE_IRQ = 7,
+ AIC3X_GPIO1_FUNC_INPUT = 8,
+ AIC3X_GPIO1_FUNC_OUTPUT = 9,
+ AIC3X_GPIO1_FUNC_DIGITAL_MIC_MODCLK = 10,
+ AIC3X_GPIO1_FUNC_AUDIO_WORDCLK = 11,
+ AIC3X_GPIO1_FUNC_BUTTON_IRQ = 12,
+ AIC3X_GPIO1_FUNC_HEADSET_DETECT_IRQ = 13,
+ AIC3X_GPIO1_FUNC_HEADSET_DETECT_OR_BUTTON_IRQ = 14,
+ AIC3X_GPIO1_FUNC_ALL_IRQ = 16
+};
+
+enum {
+ AIC3X_GPIO2_FUNC_DISABLED = 0,
+ AIC3X_GPIO2_FUNC_HEADSET_DETECT_IRQ = 2,
+ AIC3X_GPIO2_FUNC_INPUT = 3,
+ AIC3X_GPIO2_FUNC_OUTPUT = 4,
+ AIC3X_GPIO2_FUNC_DIGITAL_MIC_INPUT = 5,
+ AIC3X_GPIO2_FUNC_AUDIO_BITCLK = 8,
+ AIC3X_GPIO2_FUNC_HEADSET_DETECT_OR_BUTTON_IRQ = 9,
+ AIC3X_GPIO2_FUNC_ALL_IRQ = 10,
+ AIC3X_GPIO2_FUNC_SHORT_CIRCUIT_OR_AGC_IRQ = 11,
+ AIC3X_GPIO2_FUNC_HEADSET_OR_BUTTON_PRESS_OR_SHORT_CIRCUIT_IRQ = 12,
+ AIC3X_GPIO2_FUNC_SHORT_CIRCUIT_IRQ = 13,
+ AIC3X_GPIO2_FUNC_AGC_NOISE_IRQ = 14,
+ AIC3X_GPIO2_FUNC_BUTTON_PRESS_IRQ = 15
+};
+
+void aic3x_set_gpio(struct snd_soc_codec *codec, int gpio, int state);
+int aic3x_get_gpio(struct snd_soc_codec *codec, int gpio);
+int aic3x_headset_detected(struct snd_soc_codec *codec);
+
struct aic3x_setup_data {
unsigned short i2c_address;
+ unsigned int gpio_func[2];
};
-extern struct snd_soc_codec_dai aic3x_dai;
+extern struct snd_soc_dai aic3x_dai;
extern struct snd_soc_codec_device soc_codec_dev_aic3x;
#endif /* _AIC3X_H */
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
new file mode 100644
index 00000000000..a52d6d9e007
--- /dev/null
+++ b/sound/soc/codecs/uda1380.c
@@ -0,0 +1,852 @@
+/*
+ * uda1380.c - Philips UDA1380 ALSA SoC audio driver
+ *
+ * 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.
+ *
+ * Copyright (c) 2007 Philipp Zabel <philipp.zabel@gmail.com>
+ * Improved support for DAPM and audio routing/mixing capabilities,
+ * added TLV support.
+ *
+ * Modified by Richard Purdie <richard@openedhand.com> to fit into SoC
+ * codec model.
+ *
+ * Copyright (c) 2005 Giorgio Padrin <giorgio@mandarinlogiq.org>
+ * Copyright 2005 Openedhand Ltd.
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/types.h>
+#include <linux/string.h>
+#include <linux/slab.h>
+#include <linux/errno.h>
+#include <linux/ioctl.h>
+#include <linux/delay.h>
+#include <linux/i2c.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/initval.h>
+#include <sound/info.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+
+#include "uda1380.h"
+
+#define UDA1380_VERSION "0.6"
+#define AUDIO_NAME "uda1380"
+
+/*
+ * uda1380 register cache
+ */
+static const u16 uda1380_reg[UDA1380_CACHEREGNUM] = {
+ 0x0502, 0x0000, 0x0000, 0x3f3f,
+ 0x0202, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0xff00, 0x0000, 0x4800,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x8000, 0x0002, 0x0000,
+};
+
+/*
+ * read uda1380 register cache
+ */
+static inline unsigned int uda1380_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg == UDA1380_RESET)
+ return 0;
+ if (reg >= UDA1380_CACHEREGNUM)
+ return -1;
+ return cache[reg];
+}
+
+/*
+ * write uda1380 register cache
+ */
+static inline void uda1380_write_reg_cache(struct snd_soc_codec *codec,
+ u16 reg, unsigned int value)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg >= UDA1380_CACHEREGNUM)
+ return;
+ cache[reg] = value;
+}
+
+/*
+ * write to the UDA1380 register space
+ */
+static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[3];
+
+ /* data is
+ * data[0] is register offset
+ * data[1] is MS byte
+ * data[2] is LS byte
+ */
+ data[0] = reg;
+ data[1] = (value & 0xff00) >> 8;
+ data[2] = value & 0x00ff;
+
+ uda1380_write_reg_cache(codec, reg, value);
+
+ /* the interpolator & decimator regs must only be written when the
+ * codec DAI is active.
+ */
+ if (!codec->active && (reg >= UDA1380_MVOL))
+ return 0;
+ pr_debug("uda1380: hw write %x val %x\n", reg, value);
+ if (codec->hw_write(codec->control_data, data, 3) == 3) {
+ unsigned int val;
+ i2c_master_send(codec->control_data, data, 1);
+ i2c_master_recv(codec->control_data, data, 2);
+ val = (data[0]<<8) | data[1];
+ if (val != value) {
+ pr_debug("uda1380: READ BACK VAL %x\n",
+ (data[0]<<8) | data[1]);
+ return -EIO;
+ }
+ return 0;
+ } else
+ return -EIO;
+}
+
+#define uda1380_reset(c) uda1380_write(c, UDA1380_RESET, 0)
+
+/* declarations of ALSA reg_elem_REAL controls */
+static const char *uda1380_deemp[] = {
+ "None",
+ "32kHz",
+ "44.1kHz",
+ "48kHz",
+ "96kHz",
+};
+static const char *uda1380_input_sel[] = {
+ "Line",
+ "Mic + Line R",
+ "Line L",
+ "Mic",
+};
+static const char *uda1380_output_sel[] = {
+ "DAC",
+ "Analog Mixer",
+};
+static const char *uda1380_spf_mode[] = {
+ "Flat",
+ "Minimum1",
+ "Minimum2",
+ "Maximum"
+};
+static const char *uda1380_capture_sel[] = {
+ "ADC",
+ "Digital Mixer"
+};
+static const char *uda1380_sel_ns[] = {
+ "3rd-order",
+ "5th-order"
+};
+static const char *uda1380_mix_control[] = {
+ "off",
+ "PCM only",
+ "before sound processing",
+ "after sound processing"
+};
+static const char *uda1380_sdet_setting[] = {
+ "3200",
+ "4800",
+ "9600",
+ "19200"
+};
+static const char *uda1380_os_setting[] = {
+ "single-speed",
+ "double-speed (no mixing)",
+ "quad-speed (no mixing)"
+};
+
+static const struct soc_enum uda1380_deemp_enum[] = {
+ SOC_ENUM_SINGLE(UDA1380_DEEMP, 8, 5, uda1380_deemp),
+ SOC_ENUM_SINGLE(UDA1380_DEEMP, 0, 5, uda1380_deemp),
+};
+static const struct soc_enum uda1380_input_sel_enum =
+ SOC_ENUM_SINGLE(UDA1380_ADC, 2, 4, uda1380_input_sel); /* SEL_MIC, SEL_LNA */
+static const struct soc_enum uda1380_output_sel_enum =
+ SOC_ENUM_SINGLE(UDA1380_PM, 7, 2, uda1380_output_sel); /* R02_EN_AVC */
+static const struct soc_enum uda1380_spf_enum =
+ SOC_ENUM_SINGLE(UDA1380_MODE, 14, 4, uda1380_spf_mode); /* M */
+static const struct soc_enum uda1380_capture_sel_enum =
+ SOC_ENUM_SINGLE(UDA1380_IFACE, 6, 2, uda1380_capture_sel); /* SEL_SOURCE */
+static const struct soc_enum uda1380_sel_ns_enum =
+ SOC_ENUM_SINGLE(UDA1380_MIXER, 14, 2, uda1380_sel_ns); /* SEL_NS */
+static const struct soc_enum uda1380_mix_enum =
+ SOC_ENUM_SINGLE(UDA1380_MIXER, 12, 4, uda1380_mix_control); /* MIX, MIX_POS */
+static const struct soc_enum uda1380_sdet_enum =
+ SOC_ENUM_SINGLE(UDA1380_MIXER, 4, 4, uda1380_sdet_setting); /* SD_VALUE */
+static const struct soc_enum uda1380_os_enum =
+ SOC_ENUM_SINGLE(UDA1380_MIXER, 0, 3, uda1380_os_setting); /* OS */
+
+/*
+ * from -48 dB in 1.5 dB steps (mute instead of -49.5 dB)
+ */
+static DECLARE_TLV_DB_SCALE(amix_tlv, -4950, 150, 1);
+
+/*
+ * from -78 dB in 1 dB steps (3 dB steps, really. LSB are ignored),
+ * from -66 dB in 0.5 dB steps (2 dB steps, really) and
+ * from -52 dB in 0.25 dB steps
+ */
+static const unsigned int mvol_tlv[] = {
+ TLV_DB_RANGE_HEAD(3),
+ 0, 15, TLV_DB_SCALE_ITEM(-8200, 100, 1),
+ 16, 43, TLV_DB_SCALE_ITEM(-6600, 50, 0),
+ 44, 252, TLV_DB_SCALE_ITEM(-5200, 25, 0),
+};
+
+/*
+ * from -72 dB in 1.5 dB steps (6 dB steps really),
+ * from -66 dB in 0.75 dB steps (3 dB steps really),
+ * from -60 dB in 0.5 dB steps (2 dB steps really) and
+ * from -46 dB in 0.25 dB steps
+ */
+static const unsigned int vc_tlv[] = {
+ TLV_DB_RANGE_HEAD(4),
+ 0, 7, TLV_DB_SCALE_ITEM(-7800, 150, 1),
+ 8, 15, TLV_DB_SCALE_ITEM(-6600, 75, 0),
+ 16, 43, TLV_DB_SCALE_ITEM(-6000, 50, 0),
+ 44, 228, TLV_DB_SCALE_ITEM(-4600, 25, 0),
+};
+
+/* from 0 to 6 dB in 2 dB steps if SPF mode != flat */
+static DECLARE_TLV_DB_SCALE(tr_tlv, 0, 200, 0);
+
+/* from 0 to 24 dB in 2 dB steps, if SPF mode == maximum, otherwise cuts
+ * off at 18 dB max) */
+static DECLARE_TLV_DB_SCALE(bb_tlv, 0, 200, 0);
+
+/* from -63 to 24 dB in 0.5 dB steps (-128...48) */
+static DECLARE_TLV_DB_SCALE(dec_tlv, -6400, 50, 1);
+
+/* from 0 to 24 dB in 3 dB steps */
+static DECLARE_TLV_DB_SCALE(pga_tlv, 0, 300, 0);
+
+/* from 0 to 30 dB in 2 dB steps */
+static DECLARE_TLV_DB_SCALE(vga_tlv, 0, 200, 0);
+
+static const struct snd_kcontrol_new uda1380_snd_controls[] = {
+ SOC_DOUBLE_TLV("Analog Mixer Volume", UDA1380_AMIX, 0, 8, 44, 1, amix_tlv), /* AVCR, AVCL */
+ SOC_DOUBLE_TLV("Master Playback Volume", UDA1380_MVOL, 0, 8, 252, 1, mvol_tlv), /* MVCL, MVCR */
+ SOC_SINGLE_TLV("ADC Playback Volume", UDA1380_MIXVOL, 8, 228, 1, vc_tlv), /* VC2 */
+ SOC_SINGLE_TLV("PCM Playback Volume", UDA1380_MIXVOL, 0, 228, 1, vc_tlv), /* VC1 */
+ SOC_ENUM("Sound Processing Filter", uda1380_spf_enum), /* M */
+ SOC_DOUBLE_TLV("Tone Control - Treble", UDA1380_MODE, 4, 12, 3, 0, tr_tlv), /* TRL, TRR */
+ SOC_DOUBLE_TLV("Tone Control - Bass", UDA1380_MODE, 0, 8, 15, 0, bb_tlv), /* BBL, BBR */
+/**/ SOC_SINGLE("Master Playback Switch", UDA1380_DEEMP, 14, 1, 1), /* MTM */
+ SOC_SINGLE("ADC Playback Switch", UDA1380_DEEMP, 11, 1, 1), /* MT2 from decimation filter */
+ SOC_ENUM("ADC Playback De-emphasis", uda1380_deemp_enum[0]), /* DE2 */
+ SOC_SINGLE("PCM Playback Switch", UDA1380_DEEMP, 3, 1, 1), /* MT1, from digital data input */
+ SOC_ENUM("PCM Playback De-emphasis", uda1380_deemp_enum[1]), /* DE1 */
+ SOC_SINGLE("DAC Polarity inverting Switch", UDA1380_MIXER, 15, 1, 0), /* DA_POL_INV */
+ SOC_ENUM("Noise Shaper", uda1380_sel_ns_enum), /* SEL_NS */
+ SOC_ENUM("Digital Mixer Signal Control", uda1380_mix_enum), /* MIX_POS, MIX */
+ SOC_SINGLE("Silence Switch", UDA1380_MIXER, 7, 1, 0), /* SILENCE, force DAC output to silence */
+ SOC_SINGLE("Silence Detector Switch", UDA1380_MIXER, 6, 1, 0), /* SDET_ON */
+ SOC_ENUM("Silence Detector Setting", uda1380_sdet_enum), /* SD_VALUE */
+ SOC_ENUM("Oversampling Input", uda1380_os_enum), /* OS */
+ SOC_DOUBLE_S8_TLV("ADC Capture Volume", UDA1380_DEC, -128, 48, dec_tlv), /* ML_DEC, MR_DEC */
+/**/ SOC_SINGLE("ADC Capture Switch", UDA1380_PGA, 15, 1, 1), /* MT_ADC */
+ SOC_DOUBLE_TLV("Line Capture Volume", UDA1380_PGA, 0, 8, 8, 0, pga_tlv), /* PGA_GAINCTRLL, PGA_GAINCTRLR */
+ SOC_SINGLE("ADC Polarity inverting Switch", UDA1380_ADC, 12, 1, 0), /* ADCPOL_INV */
+ SOC_SINGLE_TLV("Mic Capture Volume", UDA1380_ADC, 8, 15, 0, vga_tlv), /* VGA_CTRL */
+ SOC_SINGLE("DC Filter Bypass Switch", UDA1380_ADC, 1, 1, 0), /* SKIP_DCFIL (before decimator) */
+ SOC_SINGLE("DC Filter Enable Switch", UDA1380_ADC, 0, 1, 0), /* EN_DCFIL (at output of decimator) */
+ SOC_SINGLE("AGC Timing", UDA1380_AGC, 8, 7, 0), /* TODO: enum, see table 62 */
+ SOC_SINGLE("AGC Target level", UDA1380_AGC, 2, 3, 1), /* AGC_LEVEL */
+ /* -5.5, -8, -11.5, -14 dBFS */
+ SOC_SINGLE("AGC Switch", UDA1380_AGC, 0, 1, 0),
+};
+
+/* add non dapm controls */
+static int uda1380_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(uda1380_snd_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&uda1380_snd_controls[i], codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+/* Input mux */
+static const struct snd_kcontrol_new uda1380_input_mux_control =
+ SOC_DAPM_ENUM("Route", uda1380_input_sel_enum);
+
+/* Output mux */
+static const struct snd_kcontrol_new uda1380_output_mux_control =
+ SOC_DAPM_ENUM("Route", uda1380_output_sel_enum);
+
+/* Capture mux */
+static const struct snd_kcontrol_new uda1380_capture_mux_control =
+ SOC_DAPM_ENUM("Route", uda1380_capture_sel_enum);
+
+
+static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
+ SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0,
+ &uda1380_input_mux_control),
+ SND_SOC_DAPM_MUX("Output Mux", SND_SOC_NOPM, 0, 0,
+ &uda1380_output_mux_control),
+ SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0,
+ &uda1380_capture_mux_control),
+ SND_SOC_DAPM_PGA("Left PGA", UDA1380_PM, 3, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Right PGA", UDA1380_PM, 1, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Mic LNA", UDA1380_PM, 4, 0, NULL, 0),
+ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", UDA1380_PM, 2, 0),
+ SND_SOC_DAPM_ADC("Right ADC", "Right Capture", UDA1380_PM, 0, 0),
+ SND_SOC_DAPM_INPUT("VINM"),
+ SND_SOC_DAPM_INPUT("VINL"),
+ SND_SOC_DAPM_INPUT("VINR"),
+ SND_SOC_DAPM_MIXER("Analog Mixer", UDA1380_PM, 6, 0, NULL, 0),
+ SND_SOC_DAPM_OUTPUT("VOUTLHP"),
+ SND_SOC_DAPM_OUTPUT("VOUTRHP"),
+ SND_SOC_DAPM_OUTPUT("VOUTL"),
+ SND_SOC_DAPM_OUTPUT("VOUTR"),
+ SND_SOC_DAPM_DAC("DAC", "Playback", UDA1380_PM, 10, 0),
+ SND_SOC_DAPM_PGA("HeadPhone Driver", UDA1380_PM, 13, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+
+ /* output mux */
+ {"HeadPhone Driver", NULL, "Output Mux"},
+ {"VOUTR", NULL, "Output Mux"},
+ {"VOUTL", NULL, "Output Mux"},
+
+ {"Analog Mixer", NULL, "VINR"},
+ {"Analog Mixer", NULL, "VINL"},
+ {"Analog Mixer", NULL, "DAC"},
+
+ {"Output Mux", "DAC", "DAC"},
+ {"Output Mux", "Analog Mixer", "Analog Mixer"},
+
+ /* {"DAC", "Digital Mixer", "I2S" } */
+
+ /* headphone driver */
+ {"VOUTLHP", NULL, "HeadPhone Driver"},
+ {"VOUTRHP", NULL, "HeadPhone Driver"},
+
+ /* input mux */
+ {"Left ADC", NULL, "Input Mux"},
+ {"Input Mux", "Mic", "Mic LNA"},
+ {"Input Mux", "Mic + Line R", "Mic LNA"},
+ {"Input Mux", "Line L", "Left PGA"},
+ {"Input Mux", "Line", "Left PGA"},
+
+ /* right input */
+ {"Right ADC", "Mic + Line R", "Right PGA"},
+ {"Right ADC", "Line", "Right PGA"},
+
+ /* inputs */
+ {"Mic LNA", NULL, "VINM"},
+ {"Left PGA", NULL, "VINL"},
+ {"Right PGA", NULL, "VINR"},
+};
+
+static int uda1380_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets,
+ ARRAY_SIZE(uda1380_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+static int uda1380_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ int iface;
+
+ /* set up DAI based upon fmt */
+ iface = uda1380_read_reg_cache(codec, UDA1380_IFACE);
+ iface &= ~(R01_SFORI_MASK | R01_SIM | R01_SFORO_MASK);
+
+ /* FIXME: how to select I2S for DATAO and MSB for DATAI correctly? */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface |= R01_SFORI_I2S | R01_SFORO_I2S;
+ break;
+ case SND_SOC_DAIFMT_LSB:
+ iface |= R01_SFORI_LSB16 | R01_SFORO_I2S;
+ break;
+ case SND_SOC_DAIFMT_MSB:
+ iface |= R01_SFORI_MSB | R01_SFORO_I2S;
+ }
+
+ if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFM)
+ iface |= R01_SIM;
+
+ uda1380_write(codec, UDA1380_IFACE, iface);
+
+ return 0;
+}
+
+/*
+ * Flush reg cache
+ * We can only write the interpolator and decimator registers
+ * when the DAI is being clocked by the CPU DAI. It's up to the
+ * machine and cpu DAI driver to do this before we are called.
+ */
+static int uda1380_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int reg, reg_start, reg_end, clk;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ reg_start = UDA1380_MVOL;
+ reg_end = UDA1380_MIXER;
+ } else {
+ reg_start = UDA1380_DEC;
+ reg_end = UDA1380_AGC;
+ }
+
+ /* FIXME disable DAC_CLK */
+ clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
+ uda1380_write(codec, UDA1380_CLK, clk & ~R00_DAC_CLK);
+
+ for (reg = reg_start; reg <= reg_end; reg++) {
+ pr_debug("uda1380: flush reg %x val %x:", reg,
+ uda1380_read_reg_cache(codec, reg));
+ uda1380_write(codec, reg, uda1380_read_reg_cache(codec, reg));
+ }
+
+ /* FIXME enable DAC_CLK */
+ uda1380_write(codec, UDA1380_CLK, clk | R00_DAC_CLK);
+
+ return 0;
+}
+
+static int uda1380_pcm_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 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
+
+ /* set WSPLL power and divider if running from this clock */
+ if (clk & R00_DAC_CLK) {
+ int rate = params_rate(params);
+ u16 pm = uda1380_read_reg_cache(codec, UDA1380_PM);
+ clk &= ~0x3; /* clear SEL_LOOP_DIV */
+ switch (rate) {
+ case 6250 ... 12500:
+ clk |= 0x0;
+ break;
+ case 12501 ... 25000:
+ clk |= 0x1;
+ break;
+ case 25001 ... 50000:
+ clk |= 0x2;
+ break;
+ case 50001 ... 100000:
+ clk |= 0x3;
+ break;
+ }
+ uda1380_write(codec, UDA1380_PM, R02_PON_PLL | pm);
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ clk |= R00_EN_DAC | R00_EN_INT;
+ else
+ clk |= R00_EN_ADC | R00_EN_DEC;
+
+ uda1380_write(codec, UDA1380_CLK, clk);
+ return 0;
+}
+
+static void uda1380_pcm_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;
+ u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
+
+ /* shut down WSPLL power if running from this clock */
+ if (clk & R00_DAC_CLK) {
+ u16 pm = uda1380_read_reg_cache(codec, UDA1380_PM);
+ uda1380_write(codec, UDA1380_PM, ~R02_PON_PLL & pm);
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ clk &= ~(R00_EN_DAC | R00_EN_INT);
+ else
+ clk &= ~(R00_EN_ADC | R00_EN_DEC);
+
+ uda1380_write(codec, UDA1380_CLK, clk);
+}
+
+static int uda1380_mute(struct snd_soc_dai *codec_dai, int mute)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 mute_reg = uda1380_read_reg_cache(codec, UDA1380_DEEMP) & ~R13_MTM;
+
+ /* FIXME: mute(codec,0) is called when the magician clock is already
+ * set to WSPLL, but for some unknown reason writing to interpolator
+ * registers works only when clocked by SYSCLK */
+ u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
+ uda1380_write(codec, UDA1380_CLK, ~R00_DAC_CLK & clk);
+ if (mute)
+ uda1380_write(codec, UDA1380_DEEMP, mute_reg | R13_MTM);
+ else
+ uda1380_write(codec, UDA1380_DEEMP, mute_reg);
+ uda1380_write(codec, UDA1380_CLK, clk);
+ return 0;
+}
+
+static int uda1380_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ int pm = uda1380_read_reg_cache(codec, UDA1380_PM);
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ case SND_SOC_BIAS_PREPARE:
+ uda1380_write(codec, UDA1380_PM, R02_PON_BIAS | pm);
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ uda1380_write(codec, UDA1380_PM, R02_PON_BIAS);
+ break;
+ case SND_SOC_BIAS_OFF:
+ uda1380_write(codec, UDA1380_PM, 0x0);
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+#define UDA1380_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
+
+struct snd_soc_dai uda1380_dai[] = {
+{
+ .name = "UDA1380",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = UDA1380_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = UDA1380_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
+ .hw_params = uda1380_pcm_hw_params,
+ .shutdown = uda1380_pcm_shutdown,
+ .prepare = uda1380_pcm_prepare,
+ },
+ .dai_ops = {
+ .digital_mute = uda1380_mute,
+ .set_fmt = uda1380_set_dai_fmt,
+ },
+},
+{ /* playback only - dual interface */
+ .name = "UDA1380",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = UDA1380_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .ops = {
+ .hw_params = uda1380_pcm_hw_params,
+ .shutdown = uda1380_pcm_shutdown,
+ .prepare = uda1380_pcm_prepare,
+ },
+ .dai_ops = {
+ .digital_mute = uda1380_mute,
+ .set_fmt = uda1380_set_dai_fmt,
+ },
+},
+{ /* capture only - dual interface*/
+ .name = "UDA1380",
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = UDA1380_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .ops = {
+ .hw_params = uda1380_pcm_hw_params,
+ .shutdown = uda1380_pcm_shutdown,
+ .prepare = uda1380_pcm_prepare,
+ },
+ .dai_ops = {
+ .set_fmt = uda1380_set_dai_fmt,
+ },
+},
+};
+EXPORT_SYMBOL_GPL(uda1380_dai);
+
+static int uda1380_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;
+
+ uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int uda1380_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ int i;
+ u8 data[2];
+ u16 *cache = codec->reg_cache;
+
+ /* Sync reg_cache with the hardware */
+ for (i = 0; i < ARRAY_SIZE(uda1380_reg); i++) {
+ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
+ data[1] = cache[i] & 0x00ff;
+ codec->hw_write(codec->control_data, data, 2);
+ }
+ uda1380_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ uda1380_set_bias_level(codec, codec->suspend_bias_level);
+ return 0;
+}
+
+/*
+ * initialise the UDA1380 driver
+ * register mixer and dsp interfaces with the kernel
+ */
+static int uda1380_init(struct snd_soc_device *socdev, int dac_clk)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret = 0;
+
+ codec->name = "UDA1380";
+ codec->owner = THIS_MODULE;
+ codec->read = uda1380_read_reg_cache;
+ codec->write = uda1380_write;
+ codec->set_bias_level = uda1380_set_bias_level;
+ codec->dai = uda1380_dai;
+ codec->num_dai = ARRAY_SIZE(uda1380_dai);
+ codec->reg_cache = kmemdup(uda1380_reg, sizeof(uda1380_reg),
+ GFP_KERNEL);
+ if (codec->reg_cache == NULL)
+ return -ENOMEM;
+ codec->reg_cache_size = ARRAY_SIZE(uda1380_reg);
+ codec->reg_cache_step = 1;
+ uda1380_reset(codec);
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ pr_err("uda1380: failed to create pcms\n");
+ goto pcm_err;
+ }
+
+ /* power on device */
+ uda1380_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ /* set clock input */
+ switch (dac_clk) {
+ case UDA1380_DAC_CLK_SYSCLK:
+ uda1380_write(codec, UDA1380_CLK, 0);
+ break;
+ case UDA1380_DAC_CLK_WSPLL:
+ uda1380_write(codec, UDA1380_CLK, R00_DAC_CLK);
+ break;
+ }
+
+ /* uda1380 init */
+ uda1380_add_controls(codec);
+ uda1380_add_widgets(codec);
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
+ pr_err("uda1380: 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 *uda1380_socdev;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+
+#define I2C_DRIVERID_UDA1380 0xfefe /* liam - need a proper id */
+
+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
+
+/* Magic definition of all other variables and things */
+I2C_CLIENT_INSMOD;
+
+static struct i2c_driver uda1380_i2c_driver;
+static struct i2c_client client_template;
+
+/* If the i2c layer weren't so broken, we could pass this kind of data
+ around */
+
+static int uda1380_codec_probe(struct i2c_adapter *adap, int addr, int kind)
+{
+ struct snd_soc_device *socdev = uda1380_socdev;
+ struct uda1380_setup_data *setup = socdev->codec_data;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct i2c_client *i2c;
+ int ret;
+
+ if (addr != setup->i2c_address)
+ return -ENODEV;
+
+ client_template.adapter = adap;
+ client_template.addr = addr;
+
+ i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL);
+ if (i2c == NULL) {
+ kfree(codec);
+ return -ENOMEM;
+ }
+ i2c_set_clientdata(i2c, codec);
+ codec->control_data = i2c;
+
+ ret = i2c_attach_client(i2c);
+ if (ret < 0) {
+ pr_err("uda1380: failed to attach codec at addr %x\n", addr);
+ goto err;
+ }
+
+ ret = uda1380_init(socdev, setup->dac_clk);
+ if (ret < 0) {
+ pr_err("uda1380: failed to initialise UDA1380\n");
+ goto err;
+ }
+ return ret;
+
+err:
+ kfree(codec);
+ kfree(i2c);
+ return ret;
+}
+
+static int uda1380_i2c_detach(struct i2c_client *client)
+{
+ struct snd_soc_codec *codec = i2c_get_clientdata(client);
+ i2c_detach_client(client);
+ kfree(codec->reg_cache);
+ kfree(client);
+ return 0;
+}
+
+static int uda1380_i2c_attach(struct i2c_adapter *adap)
+{
+ return i2c_probe(adap, &addr_data, uda1380_codec_probe);
+}
+
+static struct i2c_driver uda1380_i2c_driver = {
+ .driver = {
+ .name = "UDA1380 I2C Codec",
+ .owner = THIS_MODULE,
+ },
+ .id = I2C_DRIVERID_UDA1380,
+ .attach_adapter = uda1380_i2c_attach,
+ .detach_client = uda1380_i2c_detach,
+ .command = NULL,
+};
+
+static struct i2c_client client_template = {
+ .name = "UDA1380",
+ .driver = &uda1380_i2c_driver,
+};
+#endif
+
+static int uda1380_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct uda1380_setup_data *setup;
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ pr_info("UDA1380 Audio Codec %s", UDA1380_VERSION);
+
+ setup = socdev->codec_data;
+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (codec == NULL)
+ return -ENOMEM;
+
+ socdev->codec = codec;
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ uda1380_socdev = socdev;
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ if (setup->i2c_address) {
+ normal_i2c[0] = setup->i2c_address;
+ codec->hw_write = (hw_write_t)i2c_master_send;
+ ret = i2c_add_driver(&uda1380_i2c_driver);
+ if (ret != 0)
+ printk(KERN_ERR "can't add i2c driver");
+ }
+#else
+ /* Add other interfaces here */
+#endif
+ return ret;
+}
+
+/* power down chip */
+static int uda1380_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)
+ uda1380_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(&uda1380_i2c_driver);
+#endif
+ kfree(codec);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_uda1380 = {
+ .probe = uda1380_probe,
+ .remove = uda1380_remove,
+ .suspend = uda1380_suspend,
+ .resume = uda1380_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_uda1380);
+
+MODULE_AUTHOR("Giorgio Padrin");
+MODULE_DESCRIPTION("Audio support for codec Philips UDA1380");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/uda1380.h b/sound/soc/codecs/uda1380.h
new file mode 100644
index 00000000000..50c603e2c9f
--- /dev/null
+++ b/sound/soc/codecs/uda1380.h
@@ -0,0 +1,89 @@
+/*
+ * Audio support for Philips UDA1380
+ *
+ * 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.
+ *
+ * Copyright (c) 2005 Giorgio Padrin <giorgio@mandarinlogiq.org>
+ */
+
+#ifndef _UDA1380_H
+#define _UDA1380_H
+
+#define UDA1380_CLK 0x00
+#define UDA1380_IFACE 0x01
+#define UDA1380_PM 0x02
+#define UDA1380_AMIX 0x03
+#define UDA1380_HP 0x04
+#define UDA1380_MVOL 0x10
+#define UDA1380_MIXVOL 0x11
+#define UDA1380_MODE 0x12
+#define UDA1380_DEEMP 0x13
+#define UDA1380_MIXER 0x14
+#define UDA1380_INTSTAT 0x18
+#define UDA1380_DEC 0x20
+#define UDA1380_PGA 0x21
+#define UDA1380_ADC 0x22
+#define UDA1380_AGC 0x23
+#define UDA1380_DECSTAT 0x28
+#define UDA1380_RESET 0x7f
+
+#define UDA1380_CACHEREGNUM 0x24
+
+/* Register flags */
+#define R00_EN_ADC 0x0800
+#define R00_EN_DEC 0x0400
+#define R00_EN_DAC 0x0200
+#define R00_EN_INT 0x0100
+#define R00_DAC_CLK 0x0010
+#define R01_SFORI_I2S 0x0000
+#define R01_SFORI_LSB16 0x0100
+#define R01_SFORI_LSB18 0x0200
+#define R01_SFORI_LSB20 0x0300
+#define R01_SFORI_MSB 0x0500
+#define R01_SFORI_MASK 0x0700
+#define R01_SFORO_I2S 0x0000
+#define R01_SFORO_LSB16 0x0001
+#define R01_SFORO_LSB18 0x0002
+#define R01_SFORO_LSB20 0x0003
+#define R01_SFORO_LSB24 0x0004
+#define R01_SFORO_MSB 0x0005
+#define R01_SFORO_MASK 0x0007
+#define R01_SEL_SOURCE 0x0040
+#define R01_SIM 0x0010
+#define R02_PON_PLL 0x8000
+#define R02_PON_HP 0x2000
+#define R02_PON_DAC 0x0400
+#define R02_PON_BIAS 0x0100
+#define R02_EN_AVC 0x0080
+#define R02_PON_AVC 0x0040
+#define R02_PON_LNA 0x0010
+#define R02_PON_PGAL 0x0008
+#define R02_PON_ADCL 0x0004
+#define R02_PON_PGAR 0x0002
+#define R02_PON_ADCR 0x0001
+#define R13_MTM 0x4000
+#define R14_SILENCE 0x0080
+#define R14_SDET_ON 0x0040
+#define R21_MT_ADC 0x8000
+#define R22_SEL_LNA 0x0008
+#define R22_SEL_MIC 0x0004
+#define R22_SKIP_DCFIL 0x0002
+#define R23_AGC_EN 0x0001
+
+struct uda1380_setup_data {
+ unsigned short i2c_address;
+ int dac_clk;
+#define UDA1380_DAC_CLK_SYSCLK 0
+#define UDA1380_DAC_CLK_WSPLL 1
+};
+
+#define UDA1380_DAI_DUPLEX 0 /* playback and capture on single DAI */
+#define UDA1380_DAI_PLAYBACK 1 /* playback DAI */
+#define UDA1380_DAI_CAPTURE 2 /* capture DAI */
+
+extern struct snd_soc_dai uda1380_dai[3];
+extern struct snd_soc_codec_device soc_codec_dev_uda1380;
+
+#endif /* _UDA1380_H */
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
new file mode 100644
index 00000000000..67325fd9544
--- /dev/null
+++ b/sound/soc/codecs/wm8510.c
@@ -0,0 +1,817 @@
+/*
+ * wm8510.c -- WM8510 ALSA Soc Audio driver
+ *
+ * Copyright 2006 Wolfson Microelectronics PLC.
+ *
+ * Author: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.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/initval.h>
+
+#include "wm8510.h"
+
+#define AUDIO_NAME "wm8510"
+#define WM8510_VERSION "0.6"
+
+struct snd_soc_codec_device soc_codec_dev_wm8510;
+
+/*
+ * wm8510 register cache
+ * We can't read the WM8510 register space when we are
+ * using 2 wire for device control, so we cache them instead.
+ */
+static const u16 wm8510_reg[WM8510_CACHEREGNUM] = {
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0050, 0x0000, 0x0140, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x00ff,
+ 0x0000, 0x0000, 0x0100, 0x00ff,
+ 0x0000, 0x0000, 0x012c, 0x002c,
+ 0x002c, 0x002c, 0x002c, 0x0000,
+ 0x0032, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0038, 0x000b, 0x0032, 0x0000,
+ 0x0008, 0x000c, 0x0093, 0x00e9,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0003, 0x0010, 0x0000, 0x0000,
+ 0x0000, 0x0002, 0x0001, 0x0000,
+ 0x0000, 0x0000, 0x0039, 0x0000,
+ 0x0001,
+};
+
+/*
+ * read wm8510 register cache
+ */
+static inline unsigned int wm8510_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg == WM8510_RESET)
+ return 0;
+ if (reg >= WM8510_CACHEREGNUM)
+ return -1;
+ return cache[reg];
+}
+
+/*
+ * write wm8510 register cache
+ */
+static inline void wm8510_write_reg_cache(struct snd_soc_codec *codec,
+ u16 reg, unsigned int value)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg >= WM8510_CACHEREGNUM)
+ return;
+ cache[reg] = value;
+}
+
+/*
+ * write to the WM8510 register space
+ */
+static int wm8510_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[2];
+
+ /* data is
+ * D15..D9 WM8510 register offset
+ * D8...D0 register data
+ */
+ data[0] = (reg << 1) | ((value >> 8) & 0x0001);
+ data[1] = value & 0x00ff;
+
+ wm8510_write_reg_cache(codec, reg, value);
+ if (codec->hw_write(codec->control_data, data, 2) == 2)
+ return 0;
+ else
+ return -EIO;
+}
+
+#define wm8510_reset(c) wm8510_write(c, WM8510_RESET, 0)
+
+static const char *wm8510_companding[] = { "Off", "NC", "u-law", "A-law" };
+static const char *wm8510_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" };
+static const char *wm8510_alc[] = { "ALC", "Limiter" };
+
+static const struct soc_enum wm8510_enum[] = {
+ SOC_ENUM_SINGLE(WM8510_COMP, 1, 4, wm8510_companding), /* adc */
+ SOC_ENUM_SINGLE(WM8510_COMP, 3, 4, wm8510_companding), /* dac */
+ SOC_ENUM_SINGLE(WM8510_DAC, 4, 4, wm8510_deemp),
+ SOC_ENUM_SINGLE(WM8510_ALC3, 8, 2, wm8510_alc),
+};
+
+static const struct snd_kcontrol_new wm8510_snd_controls[] = {
+
+SOC_SINGLE("Digital Loopback Switch", WM8510_COMP, 0, 1, 0),
+
+SOC_ENUM("DAC Companding", wm8510_enum[1]),
+SOC_ENUM("ADC Companding", wm8510_enum[0]),
+
+SOC_ENUM("Playback De-emphasis", wm8510_enum[2]),
+SOC_SINGLE("DAC Inversion Switch", WM8510_DAC, 0, 1, 0),
+
+SOC_SINGLE("Master Playback Volume", WM8510_DACVOL, 0, 127, 0),
+
+SOC_SINGLE("High Pass Filter Switch", WM8510_ADC, 8, 1, 0),
+SOC_SINGLE("High Pass Cut Off", WM8510_ADC, 4, 7, 0),
+SOC_SINGLE("ADC Inversion Switch", WM8510_COMP, 0, 1, 0),
+
+SOC_SINGLE("Capture Volume", WM8510_ADCVOL, 0, 127, 0),
+
+SOC_SINGLE("DAC Playback Limiter Switch", WM8510_DACLIM1, 8, 1, 0),
+SOC_SINGLE("DAC Playback Limiter Decay", WM8510_DACLIM1, 4, 15, 0),
+SOC_SINGLE("DAC Playback Limiter Attack", WM8510_DACLIM1, 0, 15, 0),
+
+SOC_SINGLE("DAC Playback Limiter Threshold", WM8510_DACLIM2, 4, 7, 0),
+SOC_SINGLE("DAC Playback Limiter Boost", WM8510_DACLIM2, 0, 15, 0),
+
+SOC_SINGLE("ALC Enable Switch", WM8510_ALC1, 8, 1, 0),
+SOC_SINGLE("ALC Capture Max Gain", WM8510_ALC1, 3, 7, 0),
+SOC_SINGLE("ALC Capture Min Gain", WM8510_ALC1, 0, 7, 0),
+
+SOC_SINGLE("ALC Capture ZC Switch", WM8510_ALC2, 8, 1, 0),
+SOC_SINGLE("ALC Capture Hold", WM8510_ALC2, 4, 7, 0),
+SOC_SINGLE("ALC Capture Target", WM8510_ALC2, 0, 15, 0),
+
+SOC_ENUM("ALC Capture Mode", wm8510_enum[3]),
+SOC_SINGLE("ALC Capture Decay", WM8510_ALC3, 4, 15, 0),
+SOC_SINGLE("ALC Capture Attack", WM8510_ALC3, 0, 15, 0),
+
+SOC_SINGLE("ALC Capture Noise Gate Switch", WM8510_NGATE, 3, 1, 0),
+SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8510_NGATE, 0, 7, 0),
+
+SOC_SINGLE("Capture PGA ZC Switch", WM8510_INPPGA, 7, 1, 0),
+SOC_SINGLE("Capture PGA Volume", WM8510_INPPGA, 0, 63, 0),
+
+SOC_SINGLE("Speaker Playback ZC Switch", WM8510_SPKVOL, 7, 1, 0),
+SOC_SINGLE("Speaker Playback Switch", WM8510_SPKVOL, 6, 1, 1),
+SOC_SINGLE("Speaker Playback Volume", WM8510_SPKVOL, 0, 63, 0),
+SOC_SINGLE("Speaker Boost", WM8510_OUTPUT, 2, 1, 0),
+
+SOC_SINGLE("Capture Boost(+20dB)", WM8510_ADCBOOST, 8, 1, 0),
+SOC_SINGLE("Mono Playback Switch", WM8510_MONOMIX, 6, 1, 1),
+};
+
+/* add non dapm controls */
+static int wm8510_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(wm8510_snd_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm8510_snd_controls[i], codec,
+ NULL));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+/* Speaker Output Mixer */
+static const struct snd_kcontrol_new wm8510_speaker_mixer_controls[] = {
+SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_SPKMIX, 1, 1, 0),
+SOC_DAPM_SINGLE("Aux Playback Switch", WM8510_SPKMIX, 5, 1, 0),
+SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_SPKMIX, 0, 1, 0),
+};
+
+/* Mono Output Mixer */
+static const struct snd_kcontrol_new wm8510_mono_mixer_controls[] = {
+SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_MONOMIX, 1, 1, 0),
+SOC_DAPM_SINGLE("Aux Playback Switch", WM8510_MONOMIX, 2, 1, 0),
+SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_MONOMIX, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8510_boost_controls[] = {
+SOC_DAPM_SINGLE("Mic PGA Switch", WM8510_INPPGA, 6, 1, 0),
+SOC_DAPM_SINGLE("Aux Volume", WM8510_ADCBOOST, 0, 7, 0),
+SOC_DAPM_SINGLE("Mic Volume", WM8510_ADCBOOST, 4, 7, 0),
+};
+
+static const struct snd_kcontrol_new wm8510_micpga_controls[] = {
+SOC_DAPM_SINGLE("MICP Switch", WM8510_INPUT, 0, 1, 0),
+SOC_DAPM_SINGLE("MICN Switch", WM8510_INPUT, 1, 1, 0),
+SOC_DAPM_SINGLE("AUX Switch", WM8510_INPUT, 2, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget wm8510_dapm_widgets[] = {
+SND_SOC_DAPM_MIXER("Speaker Mixer", WM8510_POWER3, 2, 0,
+ &wm8510_speaker_mixer_controls[0],
+ ARRAY_SIZE(wm8510_speaker_mixer_controls)),
+SND_SOC_DAPM_MIXER("Mono Mixer", WM8510_POWER3, 3, 0,
+ &wm8510_mono_mixer_controls[0],
+ ARRAY_SIZE(wm8510_mono_mixer_controls)),
+SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8510_POWER3, 0, 0),
+SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8510_POWER2, 0, 0),
+SND_SOC_DAPM_PGA("Aux Input", WM8510_POWER1, 6, 0, NULL, 0),
+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("Boost Mixer", WM8510_POWER2, 4, 0,
+ &wm8510_boost_controls[0],
+ ARRAY_SIZE(wm8510_boost_controls)),
+
+SND_SOC_DAPM_MICBIAS("Mic Bias", WM8510_POWER1, 4, 0),
+
+SND_SOC_DAPM_INPUT("MICN"),
+SND_SOC_DAPM_INPUT("MICP"),
+SND_SOC_DAPM_INPUT("AUX"),
+SND_SOC_DAPM_OUTPUT("MONOOUT"),
+SND_SOC_DAPM_OUTPUT("SPKOUTP"),
+SND_SOC_DAPM_OUTPUT("SPKOUTN"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ /* Mono output mixer */
+ {"Mono Mixer", "PCM Playback Switch", "DAC"},
+ {"Mono Mixer", "Aux Playback Switch", "Aux Input"},
+ {"Mono Mixer", "Line Bypass Switch", "Boost Mixer"},
+
+ /* Speaker output mixer */
+ {"Speaker Mixer", "PCM Playback Switch", "DAC"},
+ {"Speaker Mixer", "Aux Playback Switch", "Aux Input"},
+ {"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"},
+
+ /* Outputs */
+ {"Mono Out", NULL, "Mono Mixer"},
+ {"MONOOUT", NULL, "Mono Out"},
+ {"SpkN Out", NULL, "Speaker Mixer"},
+ {"SpkP Out", NULL, "Speaker Mixer"},
+ {"SPKOUTN", NULL, "SpkN Out"},
+ {"SPKOUTP", NULL, "SpkP Out"},
+
+ /* Microphone PGA */
+ {"Mic PGA", "MICN Switch", "MICN"},
+ {"Mic PGA", "MICP Switch", "MICP"},
+ { "Mic PGA", "AUX Switch", "Aux Input" },
+
+ /* Boost Mixer */
+ {"Boost Mixer", "Mic PGA Switch", "Mic PGA"},
+ {"Boost Mixer", "Mic Volume", "MICP"},
+ {"Boost Mixer", "Aux Volume", "Aux Input"},
+
+ {"ADC", NULL, "Boost Mixer"},
+};
+
+static int wm8510_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, wm8510_dapm_widgets,
+ ARRAY_SIZE(wm8510_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+struct pll_ {
+ unsigned int pre_div:4; /* prescale - 1 */
+ unsigned int n:4;
+ unsigned int k;
+};
+
+static struct pll_ pll_div;
+
+/* The size in bits of the pll divide multiplied by 10
+ * to allow rounding later */
+#define FIXED_PLL_SIZE ((1 << 24) * 10)
+
+static void pll_factors(unsigned int target, unsigned int source)
+{
+ unsigned long long Kpart;
+ unsigned int K, Ndiv, Nmod;
+
+ Ndiv = target / source;
+ if (Ndiv < 6) {
+ source >>= 1;
+ pll_div.pre_div = 1;
+ Ndiv = target / source;
+ } else
+ pll_div.pre_div = 0;
+
+ if ((Ndiv < 6) || (Ndiv > 12))
+ printk(KERN_WARNING
+ "WM8510 N value %d outwith recommended range!d\n",
+ Ndiv);
+
+ pll_div.n = Ndiv;
+ Nmod = target % source;
+ Kpart = FIXED_PLL_SIZE * (long long)Nmod;
+
+ do_div(Kpart, source);
+
+ K = Kpart & 0xFFFFFFFF;
+
+ /* Check if we need to round */
+ if ((K % 10) >= 5)
+ K += 5;
+
+ /* Move down to proper range now rounding is done */
+ K /= 10;
+
+ pll_div.k = K;
+}
+
+static int wm8510_set_dai_pll(struct snd_soc_dai *codec_dai,
+ int pll_id, unsigned int freq_in, unsigned int freq_out)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 reg;
+
+ if (freq_in == 0 || freq_out == 0) {
+ /* Clock CODEC directly from MCLK */
+ reg = wm8510_read_reg_cache(codec, WM8510_CLOCK);
+ wm8510_write(codec, WM8510_CLOCK, reg & 0x0ff);
+
+ /* Turn off PLL */
+ reg = wm8510_read_reg_cache(codec, WM8510_POWER1);
+ wm8510_write(codec, WM8510_POWER1, reg & 0x1df);
+ return 0;
+ }
+
+ pll_factors(freq_out*8, freq_in);
+
+ wm8510_write(codec, WM8510_PLLN, (pll_div.pre_div << 4) | pll_div.n);
+ wm8510_write(codec, WM8510_PLLK1, pll_div.k >> 18);
+ wm8510_write(codec, WM8510_PLLK2, (pll_div.k >> 9) & 0x1ff);
+ wm8510_write(codec, WM8510_PLLK3, pll_div.k & 0x1ff);
+ reg = wm8510_read_reg_cache(codec, WM8510_POWER1);
+ wm8510_write(codec, WM8510_POWER1, reg | 0x020);
+
+ /* Run CODEC from PLL instead of MCLK */
+ reg = wm8510_read_reg_cache(codec, WM8510_CLOCK);
+ wm8510_write(codec, WM8510_CLOCK, reg | 0x100);
+
+ return 0;
+}
+
+/*
+ * Configure WM8510 clock dividers.
+ */
+static int wm8510_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
+ int div_id, int div)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 reg;
+
+ switch (div_id) {
+ case WM8510_OPCLKDIV:
+ reg = wm8510_read_reg_cache(codec, WM8510_GPIO) & 0x1cf;
+ wm8510_write(codec, WM8510_GPIO, reg | div);
+ break;
+ case WM8510_MCLKDIV:
+ reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1f;
+ wm8510_write(codec, WM8510_CLOCK, reg | div);
+ break;
+ case WM8510_ADCCLK:
+ reg = wm8510_read_reg_cache(codec, WM8510_ADC) & 0x1f7;
+ wm8510_write(codec, WM8510_ADC, reg | div);
+ break;
+ case WM8510_DACCLK:
+ reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0x1f7;
+ wm8510_write(codec, WM8510_DAC, reg | div);
+ break;
+ case WM8510_BCLKDIV:
+ reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1e3;
+ wm8510_write(codec, WM8510_CLOCK, reg | div);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int wm8510_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 iface = 0;
+ u16 clk = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1fe;
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ clk |= 0x0001;
+ 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 |= 0x0010;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface |= 0x0008;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ iface |= 0x00018;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ iface |= 0x0180;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ iface |= 0x0100;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ iface |= 0x0080;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ wm8510_write(codec, WM8510_IFACE, iface);
+ wm8510_write(codec, WM8510_CLOCK, clk);
+ return 0;
+}
+
+static int wm8510_pcm_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 = wm8510_read_reg_cache(codec, WM8510_IFACE) & 0x19f;
+ u16 adn = wm8510_read_reg_cache(codec, WM8510_ADD) & 0x1f1;
+
+ /* bit size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ iface |= 0x0020;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ iface |= 0x0040;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ iface |= 0x0060;
+ break;
+ }
+
+ /* filter coefficient */
+ switch (params_rate(params)) {
+ case SNDRV_PCM_RATE_8000:
+ adn |= 0x5 << 1;
+ break;
+ case SNDRV_PCM_RATE_11025:
+ adn |= 0x4 << 1;
+ break;
+ case SNDRV_PCM_RATE_16000:
+ adn |= 0x3 << 1;
+ break;
+ case SNDRV_PCM_RATE_22050:
+ adn |= 0x2 << 1;
+ break;
+ case SNDRV_PCM_RATE_32000:
+ adn |= 0x1 << 1;
+ break;
+ case SNDRV_PCM_RATE_44100:
+ case SNDRV_PCM_RATE_48000:
+ break;
+ }
+
+ wm8510_write(codec, WM8510_IFACE, iface);
+ wm8510_write(codec, WM8510_ADD, adn);
+ return 0;
+}
+
+static int wm8510_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 mute_reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0xffbf;
+
+ if (mute)
+ wm8510_write(codec, WM8510_DAC, mute_reg | 0x40);
+ else
+ wm8510_write(codec, WM8510_DAC, mute_reg);
+ return 0;
+}
+
+/* liam need to make this lower power with dapm */
+static int wm8510_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+
+ 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:
+ case SND_SOC_BIAS_STANDBY:
+ 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);
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+#define WM8510_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
+
+#define WM8510_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+struct snd_soc_dai wm8510_dai = {
+ .name = "WM8510 HiFi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = WM8510_RATES,
+ .formats = WM8510_FORMATS,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = WM8510_RATES,
+ .formats = WM8510_FORMATS,},
+ .ops = {
+ .hw_params = wm8510_pcm_hw_params,
+ },
+ .dai_ops = {
+ .digital_mute = wm8510_mute,
+ .set_fmt = wm8510_set_dai_fmt,
+ .set_clkdiv = wm8510_set_dai_clkdiv,
+ .set_pll = wm8510_set_dai_pll,
+ },
+};
+EXPORT_SYMBOL_GPL(wm8510_dai);
+
+static int wm8510_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;
+
+ wm8510_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int wm8510_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ int i;
+ u8 data[2];
+ u16 *cache = codec->reg_cache;
+
+ /* Sync reg_cache with the hardware */
+ for (i = 0; i < ARRAY_SIZE(wm8510_reg); i++) {
+ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
+ data[1] = cache[i] & 0x00ff;
+ codec->hw_write(codec->control_data, data, 2);
+ }
+ wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ wm8510_set_bias_level(codec, codec->suspend_bias_level);
+ return 0;
+}
+
+/*
+ * initialise the WM8510 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int wm8510_init(struct snd_soc_device *socdev)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret = 0;
+
+ codec->name = "WM8510";
+ codec->owner = THIS_MODULE;
+ codec->read = wm8510_read_reg_cache;
+ codec->write = wm8510_write;
+ codec->set_bias_level = wm8510_set_bias_level;
+ codec->dai = &wm8510_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = ARRAY_SIZE(wm8510_reg);
+ codec->reg_cache = kmemdup(wm8510_reg, sizeof(wm8510_reg), GFP_KERNEL);
+
+ if (codec->reg_cache == NULL)
+ return -ENOMEM;
+
+ wm8510_reset(codec);
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ printk(KERN_ERR "wm8510: failed to create pcms\n");
+ goto pcm_err;
+ }
+
+ /* power on device */
+ wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ wm8510_add_controls(codec);
+ wm8510_add_widgets(codec);
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "wm8510: 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 *wm8510_socdev;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+
+/*
+ * WM8510 2 wire address is 0x1a
+ */
+#define I2C_DRIVERID_WM8510 0xfefe /* liam - need a proper id */
+
+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
+
+/* Magic definition of all other variables and things */
+I2C_CLIENT_INSMOD;
+
+static struct i2c_driver wm8510_i2c_driver;
+static struct i2c_client client_template;
+
+/* If the i2c layer weren't so broken, we could pass this kind of data
+ around */
+
+static int wm8510_codec_probe(struct i2c_adapter *adap, int addr, int kind)
+{
+ struct snd_soc_device *socdev = wm8510_socdev;
+ struct wm8510_setup_data *setup = socdev->codec_data;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct i2c_client *i2c;
+ int ret;
+
+ if (addr != setup->i2c_address)
+ return -ENODEV;
+
+ client_template.adapter = adap;
+ client_template.addr = addr;
+
+ i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL);
+ if (i2c == NULL) {
+ kfree(codec);
+ return -ENOMEM;
+ }
+ i2c_set_clientdata(i2c, codec);
+ codec->control_data = i2c;
+
+ ret = i2c_attach_client(i2c);
+ if (ret < 0) {
+ pr_err("failed to attach codec at addr %x\n", addr);
+ goto err;
+ }
+
+ ret = wm8510_init(socdev);
+ if (ret < 0) {
+ pr_err("failed to initialise WM8510\n");
+ goto err;
+ }
+ return ret;
+
+err:
+ kfree(codec);
+ kfree(i2c);
+ return ret;
+}
+
+static int wm8510_i2c_detach(struct i2c_client *client)
+{
+ struct snd_soc_codec *codec = i2c_get_clientdata(client);
+ i2c_detach_client(client);
+ kfree(codec->reg_cache);
+ kfree(client);
+ return 0;
+}
+
+static int wm8510_i2c_attach(struct i2c_adapter *adap)
+{
+ return i2c_probe(adap, &addr_data, wm8510_codec_probe);
+}
+
+/* corgi i2c codec control layer */
+static struct i2c_driver wm8510_i2c_driver = {
+ .driver = {
+ .name = "WM8510 I2C Codec",
+ .owner = THIS_MODULE,
+ },
+ .id = I2C_DRIVERID_WM8510,
+ .attach_adapter = wm8510_i2c_attach,
+ .detach_client = wm8510_i2c_detach,
+ .command = NULL,
+};
+
+static struct i2c_client client_template = {
+ .name = "WM8510",
+ .driver = &wm8510_i2c_driver,
+};
+#endif
+
+static int wm8510_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct wm8510_setup_data *setup;
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ pr_info("WM8510 Audio Codec %s", WM8510_VERSION);
+
+ setup = socdev->codec_data;
+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (codec == NULL)
+ return -ENOMEM;
+
+ socdev->codec = codec;
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ wm8510_socdev = socdev;
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ if (setup->i2c_address) {
+ normal_i2c[0] = setup->i2c_address;
+ codec->hw_write = (hw_write_t)i2c_master_send;
+ ret = i2c_add_driver(&wm8510_i2c_driver);
+ if (ret != 0)
+ printk(KERN_ERR "can't add i2c driver");
+ }
+#else
+ /* Add other interfaces here */
+#endif
+ return ret;
+}
+
+/* power down chip */
+static int wm8510_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)
+ wm8510_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(&wm8510_i2c_driver);
+#endif
+ kfree(codec);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8510 = {
+ .probe = wm8510_probe,
+ .remove = wm8510_remove,
+ .suspend = wm8510_suspend,
+ .resume = wm8510_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8510);
+
+MODULE_DESCRIPTION("ASoC WM8510 driver");
+MODULE_AUTHOR("Liam Girdwood");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8510.h b/sound/soc/codecs/wm8510.h
new file mode 100644
index 00000000000..f5d2e42eb3f
--- /dev/null
+++ b/sound/soc/codecs/wm8510.h
@@ -0,0 +1,103 @@
+/*
+ * wm8510.h -- WM8510 Soc Audio driver
+ *
+ * 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 _WM8510_H
+#define _WM8510_H
+
+/* WM8510 register space */
+
+#define WM8510_RESET 0x0
+#define WM8510_POWER1 0x1
+#define WM8510_POWER2 0x2
+#define WM8510_POWER3 0x3
+#define WM8510_IFACE 0x4
+#define WM8510_COMP 0x5
+#define WM8510_CLOCK 0x6
+#define WM8510_ADD 0x7
+#define WM8510_GPIO 0x8
+#define WM8510_DAC 0xa
+#define WM8510_DACVOL 0xb
+#define WM8510_ADC 0xe
+#define WM8510_ADCVOL 0xf
+#define WM8510_EQ1 0x12
+#define WM8510_EQ2 0x13
+#define WM8510_EQ3 0x14
+#define WM8510_EQ4 0x15
+#define WM8510_EQ5 0x16
+#define WM8510_DACLIM1 0x18
+#define WM8510_DACLIM2 0x19
+#define WM8510_NOTCH1 0x1b
+#define WM8510_NOTCH2 0x1c
+#define WM8510_NOTCH3 0x1d
+#define WM8510_NOTCH4 0x1e
+#define WM8510_ALC1 0x20
+#define WM8510_ALC2 0x21
+#define WM8510_ALC3 0x22
+#define WM8510_NGATE 0x23
+#define WM8510_PLLN 0x24
+#define WM8510_PLLK1 0x25
+#define WM8510_PLLK2 0x26
+#define WM8510_PLLK3 0x27
+#define WM8510_ATTEN 0x28
+#define WM8510_INPUT 0x2c
+#define WM8510_INPPGA 0x2d
+#define WM8510_ADCBOOST 0x2f
+#define WM8510_OUTPUT 0x31
+#define WM8510_SPKMIX 0x32
+#define WM8510_SPKVOL 0x36
+#define WM8510_MONOMIX 0x38
+
+#define WM8510_CACHEREGNUM 57
+
+/* Clock divider Id's */
+#define WM8510_OPCLKDIV 0
+#define WM8510_MCLKDIV 1
+#define WM8510_ADCCLK 2
+#define WM8510_DACCLK 3
+#define WM8510_BCLKDIV 4
+
+/* DAC clock dividers */
+#define WM8510_DACCLK_F2 (1 << 3)
+#define WM8510_DACCLK_F4 (0 << 3)
+
+/* ADC clock dividers */
+#define WM8510_ADCCLK_F2 (1 << 3)
+#define WM8510_ADCCLK_F4 (0 << 3)
+
+/* PLL Out dividers */
+#define WM8510_OPCLKDIV_1 (0 << 4)
+#define WM8510_OPCLKDIV_2 (1 << 4)
+#define WM8510_OPCLKDIV_3 (2 << 4)
+#define WM8510_OPCLKDIV_4 (3 << 4)
+
+/* BCLK clock dividers */
+#define WM8510_BCLKDIV_1 (0 << 2)
+#define WM8510_BCLKDIV_2 (1 << 2)
+#define WM8510_BCLKDIV_4 (2 << 2)
+#define WM8510_BCLKDIV_8 (3 << 2)
+#define WM8510_BCLKDIV_16 (4 << 2)
+#define WM8510_BCLKDIV_32 (5 << 2)
+
+/* MCLK clock dividers */
+#define WM8510_MCLKDIV_1 (0 << 5)
+#define WM8510_MCLKDIV_1_5 (1 << 5)
+#define WM8510_MCLKDIV_2 (2 << 5)
+#define WM8510_MCLKDIV_3 (3 << 5)
+#define WM8510_MCLKDIV_4 (4 << 5)
+#define WM8510_MCLKDIV_6 (5 << 5)
+#define WM8510_MCLKDIV_8 (6 << 5)
+#define WM8510_MCLKDIV_12 (7 << 5)
+
+struct wm8510_setup_data {
+ unsigned short i2c_address;
+};
+
+extern struct snd_soc_dai wm8510_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8510;
+
+#endif
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 0cf9265fca8..369d39c3f74 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -31,25 +31,6 @@
#define AUDIO_NAME "wm8731"
#define WM8731_VERSION "0.13"
-/*
- * Debug
- */
-
-#define WM8731_DEBUG 0
-
-#ifdef WM8731_DEBUG
-#define dbg(format, arg...) \
- printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
-#else
-#define dbg(format, arg...) do {} while (0)
-#endif
-#define err(format, arg...) \
- printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
-#define info(format, arg...) \
- printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
-#define warn(format, arg...) \
- printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
-
struct snd_soc_codec_device soc_codec_dev_wm8731;
/* codec private data */
@@ -193,7 +174,7 @@ SND_SOC_DAPM_INPUT("RLINEIN"),
SND_SOC_DAPM_INPUT("LLINEIN"),
};
-static const char *intercon[][3] = {
+static const struct snd_soc_dapm_route intercon[] = {
/* output mixer */
{"Output Mixer", "Line Bypass Switch", "Line Input"},
{"Output Mixer", "HiFi Playback Switch", "DAC"},
@@ -214,22 +195,14 @@ static const char *intercon[][3] = {
{"Line Input", NULL, "LLINEIN"},
{"Line Input", NULL, "RLINEIN"},
{"Mic Bias", NULL, "MICIN"},
-
- /* terminator */
- {NULL, NULL, NULL},
};
static int wm8731_add_widgets(struct snd_soc_codec *codec)
{
- int i;
-
- for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++)
- snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]);
+ snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets,
+ ARRAY_SIZE(wm8731_dapm_widgets));
- /* set up audio path interconnects */
- for (i = 0; intercon[i][0] != NULL; i++)
- snd_soc_dapm_connect_input(codec, intercon[i][0],
- intercon[i][1], intercon[i][2]);
+ snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
snd_soc_dapm_new_widgets(codec);
return 0;
@@ -345,7 +318,7 @@ static void wm8731_shutdown(struct snd_pcm_substream *substream)
}
}
-static int wm8731_mute(struct snd_soc_codec_dai *dai, int mute)
+static int wm8731_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
u16 mute_reg = wm8731_read_reg_cache(codec, WM8731_APDIGI) & 0xfff7;
@@ -357,7 +330,7 @@ static int wm8731_mute(struct snd_soc_codec_dai *dai, int mute)
return 0;
}
-static int wm8731_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
+static int wm8731_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;
@@ -376,7 +349,7 @@ static int wm8731_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
}
-static int wm8731_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -435,29 +408,29 @@ static int wm8731_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
return 0;
}
-static int wm8731_dapm_event(struct snd_soc_codec *codec, int event)
+static int wm8731_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
{
u16 reg = wm8731_read_reg_cache(codec, WM8731_PWR) & 0xff7f;
- switch (event) {
- case SNDRV_CTL_POWER_D0: /* full On */
+ switch (level) {
+ case SND_SOC_BIAS_ON:
/* vref/mid, osc on, dac unmute */
wm8731_write(codec, WM8731_PWR, reg);
break;
- case SNDRV_CTL_POWER_D1: /* partial On */
- case SNDRV_CTL_POWER_D2: /* partial On */
+ case SND_SOC_BIAS_PREPARE:
break;
- case SNDRV_CTL_POWER_D3hot: /* Off, with power */
+ case SND_SOC_BIAS_STANDBY:
/* everything off except vref/vmid, */
wm8731_write(codec, WM8731_PWR, reg | 0x0040);
break;
- case SNDRV_CTL_POWER_D3cold: /* Off, without power */
+ case SND_SOC_BIAS_OFF:
/* everything off, dac mute, inactive */
wm8731_write(codec, WM8731_ACTIVE, 0x0);
wm8731_write(codec, WM8731_PWR, 0xffff);
break;
}
- codec->dapm_state = event;
+ codec->bias_level = level;
return 0;
}
@@ -470,7 +443,7 @@ static int wm8731_dapm_event(struct snd_soc_codec *codec, int event)
#define WM8731_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
SNDRV_PCM_FMTBIT_S24_LE)
-struct snd_soc_codec_dai wm8731_dai = {
+struct snd_soc_dai wm8731_dai = {
.name = "WM8731",
.playback = {
.stream_name = "Playback",
@@ -503,7 +476,7 @@ static int wm8731_suspend(struct platform_device *pdev, pm_message_t state)
struct snd_soc_codec *codec = socdev->codec;
wm8731_write(codec, WM8731_ACTIVE, 0x0);
- wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+ wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
@@ -521,8 +494,8 @@ static int wm8731_resume(struct platform_device *pdev)
data[1] = cache[i] & 0x00ff;
codec->hw_write(codec->control_data, data, 2);
}
- wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
- wm8731_dapm_event(codec, codec->suspend_dapm_state);
+ wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ wm8731_set_bias_level(codec, codec->suspend_bias_level);
return 0;
}
@@ -539,10 +512,10 @@ static int wm8731_init(struct snd_soc_device *socdev)
codec->owner = THIS_MODULE;
codec->read = wm8731_read_reg_cache;
codec->write = wm8731_write;
- codec->dapm_event = wm8731_dapm_event;
+ codec->set_bias_level = wm8731_set_bias_level;
codec->dai = &wm8731_dai;
codec->num_dai = 1;
- codec->reg_cache_size = sizeof(wm8731_reg);
+ codec->reg_cache_size = ARRAY_SIZE(wm8731_reg);
codec->reg_cache = kmemdup(wm8731_reg, sizeof(wm8731_reg), GFP_KERNEL);
if (codec->reg_cache == NULL)
return -ENOMEM;
@@ -557,7 +530,7 @@ static int wm8731_init(struct snd_soc_device *socdev)
}
/* power on device */
- wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+ wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
/* set the update bits */
reg = wm8731_read_reg_cache(codec, WM8731_LOUT1V);
@@ -632,13 +605,13 @@ static int wm8731_codec_probe(struct i2c_adapter *adap, int addr, int kind)
ret = i2c_attach_client(i2c);
if (ret < 0) {
- err("failed to attach codec at addr %x\n", addr);
+ pr_err("failed to attach codec at addr %x\n", addr);
goto err;
}
ret = wm8731_init(socdev);
if (ret < 0) {
- err("failed to initialise WM8731\n");
+ pr_err("failed to initialise WM8731\n");
goto err;
}
return ret;
@@ -689,7 +662,7 @@ static int wm8731_probe(struct platform_device *pdev)
struct wm8731_priv *wm8731;
int ret = 0;
- info("WM8731 Audio Codec %s", WM8731_VERSION);
+ pr_info("WM8731 Audio Codec %s", WM8731_VERSION);
setup = socdev->codec_data;
codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
@@ -730,7 +703,7 @@ static int wm8731_remove(struct platform_device *pdev)
struct snd_soc_codec *codec = socdev->codec;
if (codec->control_data)
- wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+ wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF);
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
diff --git a/sound/soc/codecs/wm8731.h b/sound/soc/codecs/wm8731.h
index 5bcab6a7afb..99f2e3c60e3 100644
--- a/sound/soc/codecs/wm8731.h
+++ b/sound/soc/codecs/wm8731.h
@@ -38,7 +38,7 @@ struct wm8731_setup_data {
unsigned short i2c_address;
};
-extern struct snd_soc_codec_dai wm8731_dai;
+extern struct snd_soc_dai wm8731_dai;
extern struct snd_soc_codec_device soc_codec_dev_wm8731;
#endif
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 16cd5d4d5ad..e23cb09f0d1 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -31,25 +31,6 @@
#define AUDIO_NAME "WM8750"
#define WM8750_VERSION "0.12"
-/*
- * Debug
- */
-
-#define WM8750_DEBUG 0
-
-#ifdef WM8750_DEBUG
-#define dbg(format, arg...) \
- printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
-#else
-#define dbg(format, arg...) do {} while (0)
-#endif
-#define err(format, arg...) \
- printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
-#define info(format, arg...) \
- printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
-#define warn(format, arg...) \
- printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
-
/* codec private data */
struct wm8750_priv {
unsigned int sysclk;
@@ -378,7 +359,7 @@ static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
SND_SOC_DAPM_INPUT("RINPUT3"),
};
-static const char *audio_map[][3] = {
+static const struct snd_soc_dapm_route audio_map[] = {
/* left mixer */
{"Left Mixer", "Playback Switch", "Left DAC"},
{"Left Mixer", "Left Bypass Switch", "Left Line Mux"},
@@ -470,22 +451,14 @@ static const char *audio_map[][3] = {
/* ADC */
{"Left ADC", NULL, "Left ADC Mux"},
{"Right ADC", NULL, "Right ADC Mux"},
-
- /* terminator */
- {NULL, NULL, NULL},
};
static int wm8750_add_widgets(struct snd_soc_codec *codec)
{
- int i;
-
- for (i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++)
- snd_soc_dapm_new_control(codec, &wm8750_dapm_widgets[i]);
+ snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets,
+ ARRAY_SIZE(wm8750_dapm_widgets));
- /* set up audio path audio_mapnects */
- for (i = 0; audio_map[i][0] != NULL; i++)
- snd_soc_dapm_connect_input(codec, audio_map[i][0],
- audio_map[i][1], audio_map[i][2]);
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
snd_soc_dapm_new_widgets(codec);
return 0;
@@ -563,7 +536,7 @@ static inline int get_coeff(int mclk, int rate)
return -EINVAL;
}
-static int wm8750_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
+static int wm8750_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;
@@ -581,7 +554,7 @@ static int wm8750_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
return -EINVAL;
}
-static int wm8750_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8750_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -674,7 +647,7 @@ static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream,
return 0;
}
-static int wm8750_mute(struct snd_soc_codec_dai *dai, int mute)
+static int wm8750_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
u16 mute_reg = wm8750_read_reg_cache(codec, WM8750_ADCDAC) & 0xfff7;
@@ -686,29 +659,29 @@ static int wm8750_mute(struct snd_soc_codec_dai *dai, int mute)
return 0;
}
-static int wm8750_dapm_event(struct snd_soc_codec *codec, int event)
+static int wm8750_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
{
u16 pwr_reg = wm8750_read_reg_cache(codec, WM8750_PWR1) & 0xfe3e;
- switch (event) {
- case SNDRV_CTL_POWER_D0: /* full On */
+ switch (level) {
+ case SND_SOC_BIAS_ON:
/* set vmid to 50k and unmute dac */
wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x00c0);
break;
- case SNDRV_CTL_POWER_D1: /* partial On */
- case SNDRV_CTL_POWER_D2: /* partial On */
+ case SND_SOC_BIAS_PREPARE:
/* set vmid to 5k for quick power up */
wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x01c1);
break;
- case SNDRV_CTL_POWER_D3hot: /* Off, with power */
+ case SND_SOC_BIAS_STANDBY:
/* mute dac and set vmid to 500k, enable VREF */
wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x0141);
break;
- case SNDRV_CTL_POWER_D3cold: /* Off, without power */
+ case SND_SOC_BIAS_OFF:
wm8750_write(codec, WM8750_PWR1, 0x0001);
break;
}
- codec->dapm_state = event;
+ codec->bias_level = level;
return 0;
}
@@ -719,7 +692,7 @@ static int wm8750_dapm_event(struct snd_soc_codec *codec, int event)
#define WM8750_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
SNDRV_PCM_FMTBIT_S24_LE)
-struct snd_soc_codec_dai wm8750_dai = {
+struct snd_soc_dai wm8750_dai = {
.name = "WM8750",
.playback = {
.stream_name = "Playback",
@@ -748,7 +721,7 @@ static void wm8750_work(struct work_struct *work)
{
struct snd_soc_codec *codec =
container_of(work, struct snd_soc_codec, delayed_work.work);
- wm8750_dapm_event(codec, codec->dapm_state);
+ wm8750_set_bias_level(codec, codec->bias_level);
}
static int wm8750_suspend(struct platform_device *pdev, pm_message_t state)
@@ -756,7 +729,7 @@ static int wm8750_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;
- wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+ wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
@@ -777,12 +750,12 @@ static int wm8750_resume(struct platform_device *pdev)
codec->hw_write(codec->control_data, data, 2);
}
- wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+ wm8750_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
/* charge wm8750 caps */
- if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) {
- wm8750_dapm_event(codec, SNDRV_CTL_POWER_D2);
- codec->dapm_state = SNDRV_CTL_POWER_D0;
+ if (codec->suspend_bias_level == SND_SOC_BIAS_ON) {
+ wm8750_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
+ codec->bias_level = SND_SOC_BIAS_ON;
schedule_delayed_work(&codec->delayed_work,
msecs_to_jiffies(1000));
}
@@ -803,10 +776,10 @@ static int wm8750_init(struct snd_soc_device *socdev)
codec->owner = THIS_MODULE;
codec->read = wm8750_read_reg_cache;
codec->write = wm8750_write;
- codec->dapm_event = wm8750_dapm_event;
+ codec->set_bias_level = wm8750_set_bias_level;
codec->dai = &wm8750_dai;
codec->num_dai = 1;
- codec->reg_cache_size = sizeof(wm8750_reg);
+ codec->reg_cache_size = ARRAY_SIZE(wm8750_reg);
codec->reg_cache = kmemdup(wm8750_reg, sizeof(wm8750_reg), GFP_KERNEL);
if (codec->reg_cache == NULL)
return -ENOMEM;
@@ -821,8 +794,8 @@ static int wm8750_init(struct snd_soc_device *socdev)
}
/* charge output caps */
- wm8750_dapm_event(codec, SNDRV_CTL_POWER_D2);
- codec->dapm_state = SNDRV_CTL_POWER_D3hot;
+ wm8750_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
+ codec->bias_level = SND_SOC_BIAS_STANDBY;
schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(1000));
/* set the update bits */
@@ -904,13 +877,13 @@ static int wm8750_codec_probe(struct i2c_adapter *adap, int addr, int kind)
ret = i2c_attach_client(i2c);
if (ret < 0) {
- err("failed to attach codec at addr %x\n", addr);
+ pr_err("failed to attach codec at addr %x\n", addr);
goto err;
}
ret = wm8750_init(socdev);
if (ret < 0) {
- err("failed to initialise WM8750\n");
+ pr_err("failed to initialise WM8750\n");
goto err;
}
return ret;
@@ -961,7 +934,7 @@ static int wm8750_probe(struct platform_device *pdev)
struct wm8750_priv *wm8750;
int ret = 0;
- info("WM8750 Audio Codec %s", WM8750_VERSION);
+ pr_info("WM8750 Audio Codec %s", WM8750_VERSION);
codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
if (codec == NULL)
return -ENOMEM;
@@ -1021,7 +994,7 @@ static int wm8750_remove(struct platform_device *pdev)
struct snd_soc_codec *codec = socdev->codec;
if (codec->control_data)
- wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+ wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF);
run_delayed_work(&codec->delayed_work);
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
diff --git a/sound/soc/codecs/wm8750.h b/sound/soc/codecs/wm8750.h
index a97a54a6348..8ef30e628b2 100644
--- a/sound/soc/codecs/wm8750.h
+++ b/sound/soc/codecs/wm8750.h
@@ -61,7 +61,7 @@ struct wm8750_setup_data {
unsigned short i2c_address;
};
-extern struct snd_soc_codec_dai wm8750_dai;
+extern struct snd_soc_dai wm8750_dai;
extern struct snd_soc_codec_device soc_codec_dev_wm8750;
#endif
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index fb41826c4c4..8604809f0c3 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -55,25 +55,6 @@
#define AUDIO_NAME "wm8753"
#define WM8753_VERSION "0.16"
-/*
- * Debug
- */
-
-#define WM8753_DEBUG 0
-
-#ifdef WM8753_DEBUG
-#define dbg(format, arg...) \
- printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
-#else
-#define dbg(format, arg...) do {} while (0)
-#endif
-#define err(format, arg...) \
- printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
-#define info(format, arg...) \
- printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
-#define warn(format, arg...) \
- printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
-
static int caps_charge = 2000;
module_param(caps_charge, int, 0);
MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)");
@@ -260,28 +241,50 @@ static int wm8753_set_dai(struct snd_kcontrol *kcontrol,
return 1;
}
-static const DECLARE_TLV_DB_LINEAR(rec_mix_tlv, -1500, 600);
+static const DECLARE_TLV_DB_SCALE(rec_mix_tlv, -1500, 300, 0);
+static const DECLARE_TLV_DB_SCALE(mic_preamp_tlv, 1200, 600, 0);
+static const DECLARE_TLV_DB_SCALE(adc_tlv, -9750, 50, 1);
+static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1);
+static const unsigned int out_tlv[] = {
+ TLV_DB_RANGE_HEAD(2),
+ /* 0000000 - 0101111 = "Analogue mute" */
+ 0, 48, TLV_DB_SCALE_ITEM(-25500, 0, 0),
+ 48, 127, TLV_DB_SCALE_ITEM(-7300, 100, 0),
+};
+static const DECLARE_TLV_DB_SCALE(mix_tlv, -1500, 300, 0);
+static const DECLARE_TLV_DB_SCALE(voice_mix_tlv, -1200, 300, 0);
+static const DECLARE_TLV_DB_SCALE(pga_tlv, -1725, 75, 0);
static const struct snd_kcontrol_new wm8753_snd_controls[] = {
-SOC_DOUBLE_R("PCM Volume", WM8753_LDAC, WM8753_RDAC, 0, 255, 0),
-
-SOC_DOUBLE_R("ADC Capture Volume", WM8753_LADC, WM8753_RADC, 0, 255, 0),
-
-SOC_DOUBLE_R("Headphone Playback Volume", WM8753_LOUT1V, WM8753_ROUT1V, 0, 127, 0),
-SOC_DOUBLE_R("Speaker Playback Volume", WM8753_LOUT2V, WM8753_ROUT2V, 0, 127, 0),
-
-SOC_SINGLE("Mono Playback Volume", WM8753_MOUTV, 0, 127, 0),
-
-SOC_DOUBLE_R("Bypass Playback Volume", WM8753_LOUTM1, WM8753_ROUTM1, 4, 7, 1),
-SOC_DOUBLE_R("Sidetone Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 4, 7, 1),
-SOC_DOUBLE_R("Voice Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 0, 7, 1),
-
-SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8753_LOUT1V, WM8753_ROUT1V, 7, 1, 0),
-SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8753_LOUT2V, WM8753_ROUT2V, 7, 1, 0),
-
-SOC_SINGLE("Mono Bypass Playback Volume", WM8753_MOUTM1, 4, 7, 1),
-SOC_SINGLE("Mono Sidetone Playback Volume", WM8753_MOUTM2, 4, 7, 1),
-SOC_SINGLE("Mono Voice Playback Volume", WM8753_MOUTM2, 0, 7, 1),
+SOC_DOUBLE_R_TLV("PCM Volume", WM8753_LDAC, WM8753_RDAC, 0, 255, 0, dac_tlv),
+
+SOC_DOUBLE_R_TLV("ADC Capture Volume", WM8753_LADC, WM8753_RADC, 0, 255, 0,
+ adc_tlv),
+
+SOC_DOUBLE_R_TLV("Headphone Playback Volume", WM8753_LOUT1V, WM8753_ROUT1V,
+ 0, 127, 0, out_tlv),
+SOC_DOUBLE_R_TLV("Speaker Playback Volume", WM8753_LOUT2V, WM8753_ROUT2V, 0,
+ 127, 0, out_tlv),
+
+SOC_SINGLE_TLV("Mono Playback Volume", WM8753_MOUTV, 0, 127, 0, out_tlv),
+
+SOC_DOUBLE_R_TLV("Bypass Playback Volume", WM8753_LOUTM1, WM8753_ROUTM1, 4, 7,
+ 1, mix_tlv),
+SOC_DOUBLE_R_TLV("Sidetone Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 4,
+ 7, 1, mix_tlv),
+SOC_DOUBLE_R_TLV("Voice Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 0, 7,
+ 1, voice_mix_tlv),
+
+SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8753_LOUT1V, WM8753_ROUT1V, 7,
+ 1, 0),
+SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8753_LOUT2V, WM8753_ROUT2V, 7,
+ 1, 0),
+
+SOC_SINGLE_TLV("Mono Bypass Playback Volume", WM8753_MOUTM1, 4, 7, 1, mix_tlv),
+SOC_SINGLE_TLV("Mono Sidetone Playback Volume", WM8753_MOUTM2, 4, 7, 1,
+ mix_tlv),
+SOC_SINGLE_TLV("Mono Voice Playback Volume", WM8753_MOUTM2, 0, 7, 1,
+ voice_mix_tlv),
SOC_SINGLE("Mono Playback ZC Switch", WM8753_MOUTV, 7, 1, 0),
SOC_ENUM("Bass Boost", wm8753_enum[0]),
@@ -291,10 +294,13 @@ SOC_SINGLE("Bass Volume", WM8753_BASS, 0, 15, 1),
SOC_SINGLE("Treble Volume", WM8753_TREBLE, 0, 15, 1),
SOC_ENUM("Treble Cut-off", wm8753_enum[2]),
-SOC_DOUBLE_TLV("Sidetone Capture Volume", WM8753_RECMIX1, 0, 4, 7, 1, rec_mix_tlv),
-SOC_SINGLE_TLV("Voice Sidetone Capture Volume", WM8753_RECMIX2, 0, 7, 1, rec_mix_tlv),
+SOC_DOUBLE_TLV("Sidetone Capture Volume", WM8753_RECMIX1, 0, 4, 7, 1,
+ rec_mix_tlv),
+SOC_SINGLE_TLV("Voice Sidetone Capture Volume", WM8753_RECMIX2, 0, 7, 1,
+ rec_mix_tlv),
-SOC_DOUBLE_R("Capture Volume", WM8753_LINVOL, WM8753_RINVOL, 0, 63, 0),
+SOC_DOUBLE_R_TLV("Capture Volume", WM8753_LINVOL, WM8753_RINVOL, 0, 63, 0,
+ pga_tlv),
SOC_DOUBLE_R("Capture ZC Switch", WM8753_LINVOL, WM8753_RINVOL, 6, 1, 0),
SOC_DOUBLE_R("Capture Switch", WM8753_LINVOL, WM8753_RINVOL, 7, 1, 1),
@@ -326,8 +332,8 @@ SOC_ENUM("De-emphasis", wm8753_enum[8]),
SOC_ENUM("Playback Mono Mix", wm8753_enum[9]),
SOC_ENUM("Playback Phase", wm8753_enum[10]),
-SOC_SINGLE("Mic2 Capture Volume", WM8753_INCTL1, 7, 3, 0),
-SOC_SINGLE("Mic1 Capture Volume", WM8753_INCTL1, 5, 3, 0),
+SOC_SINGLE_TLV("Mic2 Capture Volume", WM8753_INCTL1, 7, 3, 0, mic_preamp_tlv),
+SOC_SINGLE_TLV("Mic1 Capture Volume", WM8753_INCTL1, 5, 3, 0, mic_preamp_tlv),
SOC_ENUM_EXT("DAI Mode", wm8753_enum[26], wm8753_get_dai, wm8753_set_dai),
@@ -523,7 +529,7 @@ SND_SOC_DAPM_INPUT("MIC2"),
SND_SOC_DAPM_VMID("VREF"),
};
-static const char *audio_map[][3] = {
+static const struct snd_soc_dapm_route audio_map[] = {
/* left mixer */
{"Left Mixer", "Left Playback Switch", "Left DAC"},
{"Left Mixer", "Voice Playback Switch", "Voice DAC"},
@@ -674,23 +680,14 @@ static const char *audio_map[][3] = {
/* ACOP */
{"ACOP", NULL, "ALC Mixer"},
-
- /* terminator */
- {NULL, NULL, NULL},
};
static int wm8753_add_widgets(struct snd_soc_codec *codec)
{
- int i;
+ snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets,
+ ARRAY_SIZE(wm8753_dapm_widgets));
- for (i = 0; i < ARRAY_SIZE(wm8753_dapm_widgets); i++)
- snd_soc_dapm_new_control(codec, &wm8753_dapm_widgets[i]);
-
- /* set up the WM8753 audio map */
- for (i = 0; audio_map[i][0] != NULL; i++) {
- snd_soc_dapm_connect_input(codec, audio_map[i][0],
- audio_map[i][1], audio_map[i][2]);
- }
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
snd_soc_dapm_new_widgets(codec);
return 0;
@@ -743,7 +740,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target,
pll_div->k = K;
}
-static int wm8753_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_set_dai_pll(struct snd_soc_dai *codec_dai,
int pll_id, unsigned int freq_in, unsigned int freq_out)
{
u16 reg, enable;
@@ -866,7 +863,7 @@ static int get_coeff(int mclk, int rate)
/*
* Clock after PLL and dividers
*/
-static int wm8753_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_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;
@@ -893,7 +890,7 @@ static int wm8753_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
/*
* Set's ADC and Voice DAC format.
*/
-static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -963,7 +960,7 @@ static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream,
/*
* Set's PCM dai fmt and BCLK.
*/
-static int wm8753_pcm_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -1029,7 +1026,7 @@ static int wm8753_pcm_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
return 0;
}
-static int wm8753_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
int div_id, int div)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -1057,7 +1054,7 @@ static int wm8753_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
/*
* Set's HiFi DAC format.
*/
-static int wm8753_hdac_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_hdac_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -1090,7 +1087,7 @@ static int wm8753_hdac_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
/*
* Set's I2S DAI format.
*/
-static int wm8753_i2s_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -1198,7 +1195,7 @@ static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream,
return 0;
}
-static int wm8753_mode1v_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_mode1v_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -1213,7 +1210,7 @@ static int wm8753_mode1v_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
return wm8753_pcm_set_dai_fmt(codec_dai, fmt);
}
-static int wm8753_mode1h_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_mode1h_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
if (wm8753_hdac_set_dai_fmt(codec_dai, fmt) < 0)
@@ -1221,7 +1218,7 @@ static int wm8753_mode1h_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
return wm8753_i2s_set_dai_fmt(codec_dai, fmt);
}
-static int wm8753_mode2_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_mode2_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -1236,7 +1233,7 @@ static int wm8753_mode2_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
return wm8753_i2s_set_dai_fmt(codec_dai, fmt);
}
-static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -1253,7 +1250,7 @@ static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
return wm8753_i2s_set_dai_fmt(codec_dai, fmt);
}
-static int wm8753_mute(struct snd_soc_codec_dai *dai, int mute)
+static int wm8753_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
u16 mute_reg = wm8753_read_reg_cache(codec, WM8753_DAC) & 0xfff7;
@@ -1274,29 +1271,29 @@ static int wm8753_mute(struct snd_soc_codec_dai *dai, int mute)
return 0;
}
-static int wm8753_dapm_event(struct snd_soc_codec *codec, int event)
+static int wm8753_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
{
u16 pwr_reg = wm8753_read_reg_cache(codec, WM8753_PWR1) & 0xfe3e;
- switch (event) {
- case SNDRV_CTL_POWER_D0: /* full On */
+ switch (level) {
+ case SND_SOC_BIAS_ON:
/* set vmid to 50k and unmute dac */
wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x00c0);
break;
- case SNDRV_CTL_POWER_D1: /* partial On */
- case SNDRV_CTL_POWER_D2: /* partial On */
+ case SND_SOC_BIAS_PREPARE:
/* set vmid to 5k for quick power up */
wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x01c1);
break;
- case SNDRV_CTL_POWER_D3hot: /* Off, with power */
+ case SND_SOC_BIAS_STANDBY:
/* mute dac and set vmid to 500k, enable VREF */
wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x0141);
break;
- case SNDRV_CTL_POWER_D3cold: /* Off, without power */
+ case SND_SOC_BIAS_OFF:
wm8753_write(codec, WM8753_PWR1, 0x0001);
break;
}
- codec->dapm_state = event;
+ codec->bias_level = level;
return 0;
}
@@ -1319,7 +1316,7 @@ static int wm8753_dapm_event(struct snd_soc_codec *codec, int event)
* 3. Voice disabled - HIFI over HIFI
* 4. Voice disabled - HIFI over HIFI, uses voice DAI LRC for capture
*/
-static const struct snd_soc_codec_dai wm8753_all_dai[] = {
+static const struct snd_soc_dai wm8753_all_dai[] = {
/* DAI HiFi mode 1 */
{ .name = "WM8753 HiFi",
.id = 1,
@@ -1459,7 +1456,7 @@ static const struct snd_soc_codec_dai wm8753_all_dai[] = {
},
};
-struct snd_soc_codec_dai wm8753_dai[2];
+struct snd_soc_dai wm8753_dai[2];
EXPORT_SYMBOL_GPL(wm8753_dai);
static void wm8753_set_dai_mode(struct snd_soc_codec *codec, unsigned int mode)
@@ -1500,7 +1497,7 @@ static void wm8753_work(struct work_struct *work)
{
struct snd_soc_codec *codec =
container_of(work, struct snd_soc_codec, delayed_work.work);
- wm8753_dapm_event(codec, codec->dapm_state);
+ wm8753_set_bias_level(codec, codec->bias_level);
}
static int wm8753_suspend(struct platform_device *pdev, pm_message_t state)
@@ -1512,7 +1509,7 @@ static int wm8753_suspend(struct platform_device *pdev, pm_message_t state)
if (!codec->card)
return 0;
- wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+ wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
@@ -1537,12 +1534,12 @@ static int wm8753_resume(struct platform_device *pdev)
codec->hw_write(codec->control_data, data, 2);
}
- wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+ wm8753_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
/* charge wm8753 caps */
- if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) {
- wm8753_dapm_event(codec, SNDRV_CTL_POWER_D2);
- codec->dapm_state = SNDRV_CTL_POWER_D0;
+ if (codec->suspend_bias_level == SND_SOC_BIAS_ON) {
+ wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
+ codec->bias_level = SND_SOC_BIAS_ON;
schedule_delayed_work(&codec->delayed_work,
msecs_to_jiffies(caps_charge));
}
@@ -1563,10 +1560,10 @@ static int wm8753_init(struct snd_soc_device *socdev)
codec->owner = THIS_MODULE;
codec->read = wm8753_read_reg_cache;
codec->write = wm8753_write;
- codec->dapm_event = wm8753_dapm_event;
+ codec->set_bias_level = wm8753_set_bias_level;
codec->dai = wm8753_dai;
codec->num_dai = 2;
- codec->reg_cache_size = sizeof(wm8753_reg);
+ codec->reg_cache_size = ARRAY_SIZE(wm8753_reg);
codec->reg_cache = kmemdup(wm8753_reg, sizeof(wm8753_reg), GFP_KERNEL);
if (codec->reg_cache == NULL)
@@ -1584,8 +1581,8 @@ static int wm8753_init(struct snd_soc_device *socdev)
}
/* charge output caps */
- wm8753_dapm_event(codec, SNDRV_CTL_POWER_D2);
- codec->dapm_state = SNDRV_CTL_POWER_D3hot;
+ wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
+ codec->bias_level = SND_SOC_BIAS_STANDBY;
schedule_delayed_work(&codec->delayed_work,
msecs_to_jiffies(caps_charge));
@@ -1673,13 +1670,13 @@ static int wm8753_codec_probe(struct i2c_adapter *adap, int addr, int kind)
ret = i2c_attach_client(i2c);
if (ret < 0) {
- err("failed to attach codec at addr %x\n", addr);
+ pr_err("failed to attach codec at addr %x\n", addr);
goto err;
}
ret = wm8753_init(socdev);
if (ret < 0) {
- err("failed to initialise WM8753\n");
+ pr_err("failed to initialise WM8753\n");
goto err;
}
@@ -1731,7 +1728,7 @@ static int wm8753_probe(struct platform_device *pdev)
struct wm8753_priv *wm8753;
int ret = 0;
- info("WM8753 Audio Codec %s", WM8753_VERSION);
+ pr_info("WM8753 Audio Codec %s", WM8753_VERSION);
setup = socdev->codec_data;
codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
@@ -1792,7 +1789,7 @@ static int wm8753_remove(struct platform_device *pdev)
struct snd_soc_codec *codec = socdev->codec;
if (codec->control_data)
- wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+ wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF);
run_delayed_work(&codec->delayed_work);
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
diff --git a/sound/soc/codecs/wm8753.h b/sound/soc/codecs/wm8753.h
index 95e2a1f5316..44f5f1ff0cc 100644
--- a/sound/soc/codecs/wm8753.h
+++ b/sound/soc/codecs/wm8753.h
@@ -120,7 +120,7 @@ struct wm8753_setup_data {
#define WM8753_DAI_HIFI 0
#define WM8753_DAI_VOICE 1
-extern struct snd_soc_codec_dai wm8753_dai[2];
+extern struct snd_soc_dai wm8753_dai[2];
extern struct snd_soc_codec_device soc_codec_dev_wm8753;
#endif
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
new file mode 100644
index 00000000000..3ecce5168e9
--- /dev/null
+++ b/sound/soc/codecs/wm8990.c
@@ -0,0 +1,1626 @@
+/*
+ * wm8990.c -- WM8990 ALSA Soc Audio driver
+ *
+ * Copyright 2008 Wolfson Microelectronics PLC.
+ * Author: Liam Girdwood
+ * lg@opensource.wolfsonmicro.com or linux@wolfsonmicro.com
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/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/initval.h>
+#include <sound/tlv.h>
+#include <asm/div64.h>
+
+#include "wm8990.h"
+
+#define AUDIO_NAME "wm8990"
+#define WM8990_VERSION "0.2"
+
+/* codec private data */
+struct wm8990_priv {
+ unsigned int sysclk;
+ unsigned int pcmclk;
+};
+
+/*
+ * wm8990 register cache. Note that register 0 is not included in the
+ * cache.
+ */
+static const u16 wm8990_reg[] = {
+ 0x8990, /* R0 - Reset */
+ 0x0000, /* R1 - Power Management (1) */
+ 0x6000, /* R2 - Power Management (2) */
+ 0x0000, /* R3 - Power Management (3) */
+ 0x4050, /* R4 - Audio Interface (1) */
+ 0x4000, /* R5 - Audio Interface (2) */
+ 0x01C8, /* R6 - Clocking (1) */
+ 0x0000, /* R7 - Clocking (2) */
+ 0x0040, /* R8 - Audio Interface (3) */
+ 0x0040, /* R9 - Audio Interface (4) */
+ 0x0004, /* R10 - DAC CTRL */
+ 0x00C0, /* R11 - Left DAC Digital Volume */
+ 0x00C0, /* R12 - Right DAC Digital Volume */
+ 0x0000, /* R13 - Digital Side Tone */
+ 0x0100, /* R14 - ADC CTRL */
+ 0x00C0, /* R15 - Left ADC Digital Volume */
+ 0x00C0, /* R16 - Right ADC Digital Volume */
+ 0x0000, /* R17 */
+ 0x0000, /* R18 - GPIO CTRL 1 */
+ 0x1000, /* R19 - GPIO1 & GPIO2 */
+ 0x1010, /* R20 - GPIO3 & GPIO4 */
+ 0x1010, /* R21 - GPIO5 & GPIO6 */
+ 0x8000, /* R22 - GPIOCTRL 2 */
+ 0x0800, /* R23 - GPIO_POL */
+ 0x008B, /* R24 - Left Line Input 1&2 Volume */
+ 0x008B, /* R25 - Left Line Input 3&4 Volume */
+ 0x008B, /* R26 - Right Line Input 1&2 Volume */
+ 0x008B, /* R27 - Right Line Input 3&4 Volume */
+ 0x0000, /* R28 - Left Output Volume */
+ 0x0000, /* R29 - Right Output Volume */
+ 0x0066, /* R30 - Line Outputs Volume */
+ 0x0022, /* R31 - Out3/4 Volume */
+ 0x0079, /* R32 - Left OPGA Volume */
+ 0x0079, /* R33 - Right OPGA Volume */
+ 0x0003, /* R34 - Speaker Volume */
+ 0x0003, /* R35 - ClassD1 */
+ 0x0000, /* R36 */
+ 0x0100, /* R37 - ClassD3 */
+ 0x0000, /* R38 */
+ 0x0000, /* R39 - Input Mixer1 */
+ 0x0000, /* R40 - Input Mixer2 */
+ 0x0000, /* R41 - Input Mixer3 */
+ 0x0000, /* R42 - Input Mixer4 */
+ 0x0000, /* R43 - Input Mixer5 */
+ 0x0000, /* R44 - Input Mixer6 */
+ 0x0000, /* R45 - Output Mixer1 */
+ 0x0000, /* R46 - Output Mixer2 */
+ 0x0000, /* R47 - Output Mixer3 */
+ 0x0000, /* R48 - Output Mixer4 */
+ 0x0000, /* R49 - Output Mixer5 */
+ 0x0000, /* R50 - Output Mixer6 */
+ 0x0180, /* R51 - Out3/4 Mixer */
+ 0x0000, /* R52 - Line Mixer1 */
+ 0x0000, /* R53 - Line Mixer2 */
+ 0x0000, /* R54 - Speaker Mixer */
+ 0x0000, /* R55 - Additional Control */
+ 0x0000, /* R56 - AntiPOP1 */
+ 0x0000, /* R57 - AntiPOP2 */
+ 0x0000, /* R58 - MICBIAS */
+ 0x0000, /* R59 */
+ 0x0008, /* R60 - PLL1 */
+ 0x0031, /* R61 - PLL2 */
+ 0x0026, /* R62 - PLL3 */
+};
+
+/*
+ * read wm8990 register cache
+ */
+static inline unsigned int wm8990_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+ BUG_ON(reg > (ARRAY_SIZE(wm8990_reg)) - 1);
+ return cache[reg];
+}
+
+/*
+ * write wm8990 register cache
+ */
+static inline void wm8990_write_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg, unsigned int value)
+{
+ u16 *cache = codec->reg_cache;
+ BUG_ON(reg > (ARRAY_SIZE(wm8990_reg)) - 1);
+
+ /* Reset register is uncached */
+ if (reg == 0)
+ return;
+
+ cache[reg] = value;
+}
+
+/*
+ * write to the wm8990 register space
+ */
+static int wm8990_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[3];
+
+ data[0] = reg & 0xFF;
+ data[1] = (value >> 8) & 0xFF;
+ data[2] = value & 0xFF;
+
+ wm8990_write_reg_cache(codec, reg, value);
+
+ if (codec->hw_write(codec->control_data, data, 3) == 2)
+ return 0;
+ else
+ return -EIO;
+}
+
+#define wm8990_reset(c) wm8990_write(c, WM8990_RESET, 0)
+
+static const DECLARE_TLV_DB_LINEAR(rec_mix_tlv, -1500, 600);
+
+static const DECLARE_TLV_DB_LINEAR(in_pga_tlv, -1650, 3000);
+
+static const DECLARE_TLV_DB_LINEAR(out_mix_tlv, 0, -2100);
+
+static const DECLARE_TLV_DB_LINEAR(out_pga_tlv, -7300, 600);
+
+static const DECLARE_TLV_DB_LINEAR(out_omix_tlv, -600, 0);
+
+static const DECLARE_TLV_DB_LINEAR(out_dac_tlv, -7163, 0);
+
+static const DECLARE_TLV_DB_LINEAR(in_adc_tlv, -7163, 1763);
+
+static const DECLARE_TLV_DB_LINEAR(out_sidetone_tlv, -3600, 0);
+
+static int wm899x_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xff;
+ int ret;
+ u16 val;
+
+ ret = snd_soc_put_volsw(kcontrol, ucontrol);
+ if (ret < 0)
+ return ret;
+
+ /* now hit the volume update bits (always bit 8) */
+ val = wm8990_read_reg_cache(codec, reg);
+ return wm8990_write(codec, reg, val | 0x0100);
+}
+
+#define SOC_WM899X_OUTPGA_SINGLE_R_TLV(xname, reg, shift, max, invert,\
+ tlv_array) {\
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+ SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+ .tlv.p = (tlv_array), \
+ .info = snd_soc_info_volsw, \
+ .get = snd_soc_get_volsw, .put = wm899x_outpga_put_volsw_vu, \
+ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) }
+
+
+static const char *wm8990_digital_sidetone[] =
+ {"None", "Left ADC", "Right ADC", "Reserved"};
+
+static const struct soc_enum wm8990_left_digital_sidetone_enum =
+SOC_ENUM_SINGLE(WM8990_DIGITAL_SIDE_TONE,
+ WM8990_ADC_TO_DACL_SHIFT,
+ WM8990_ADC_TO_DACL_MASK,
+ wm8990_digital_sidetone);
+
+static const struct soc_enum wm8990_right_digital_sidetone_enum =
+SOC_ENUM_SINGLE(WM8990_DIGITAL_SIDE_TONE,
+ WM8990_ADC_TO_DACR_SHIFT,
+ WM8990_ADC_TO_DACR_MASK,
+ wm8990_digital_sidetone);
+
+static const char *wm8990_adcmode[] =
+ {"Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"};
+
+static const struct soc_enum wm8990_right_adcmode_enum =
+SOC_ENUM_SINGLE(WM8990_ADC_CTRL,
+ WM8990_ADC_HPF_CUT_SHIFT,
+ WM8990_ADC_HPF_CUT_MASK,
+ wm8990_adcmode);
+
+static const struct snd_kcontrol_new wm8990_snd_controls[] = {
+/* INMIXL */
+SOC_SINGLE("LIN12 PGA Boost", WM8990_INPUT_MIXER3, WM8990_L12MNBST_BIT, 1, 0),
+SOC_SINGLE("LIN34 PGA Boost", WM8990_INPUT_MIXER3, WM8990_L34MNBST_BIT, 1, 0),
+/* INMIXR */
+SOC_SINGLE("RIN12 PGA Boost", WM8990_INPUT_MIXER3, WM8990_R12MNBST_BIT, 1, 0),
+SOC_SINGLE("RIN34 PGA Boost", WM8990_INPUT_MIXER3, WM8990_R34MNBST_BIT, 1, 0),
+
+/* LOMIX */
+SOC_SINGLE_TLV("LOMIX LIN3 Bypass Volume", WM8990_OUTPUT_MIXER3,
+ WM8990_LLI3LOVOL_SHIFT, WM8990_LLI3LOVOL_MASK, 1, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX RIN12 PGA Bypass Volume", WM8990_OUTPUT_MIXER3,
+ WM8990_LR12LOVOL_SHIFT, WM8990_LR12LOVOL_MASK, 1, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX LIN12 PGA Bypass Volume", WM8990_OUTPUT_MIXER3,
+ WM8990_LL12LOVOL_SHIFT, WM8990_LL12LOVOL_MASK, 1, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX RIN3 Bypass Volume", WM8990_OUTPUT_MIXER5,
+ WM8990_LRI3LOVOL_SHIFT, WM8990_LRI3LOVOL_MASK, 1, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX AINRMUX Bypass Volume", WM8990_OUTPUT_MIXER5,
+ WM8990_LRBLOVOL_SHIFT, WM8990_LRBLOVOL_MASK, 1, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX AINLMUX Bypass Volume", WM8990_OUTPUT_MIXER5,
+ WM8990_LRBLOVOL_SHIFT, WM8990_LRBLOVOL_MASK, 1, out_mix_tlv),
+
+/* ROMIX */
+SOC_SINGLE_TLV("ROMIX RIN3 Bypass Volume", WM8990_OUTPUT_MIXER4,
+ WM8990_RRI3ROVOL_SHIFT, WM8990_RRI3ROVOL_MASK, 1, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX LIN12 PGA Bypass Volume", WM8990_OUTPUT_MIXER4,
+ WM8990_RL12ROVOL_SHIFT, WM8990_RL12ROVOL_MASK, 1, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX RIN12 PGA Bypass Volume", WM8990_OUTPUT_MIXER4,
+ WM8990_RR12ROVOL_SHIFT, WM8990_RR12ROVOL_MASK, 1, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX LIN3 Bypass Volume", WM8990_OUTPUT_MIXER6,
+ WM8990_RLI3ROVOL_SHIFT, WM8990_RLI3ROVOL_MASK, 1, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX AINLMUX Bypass Volume", WM8990_OUTPUT_MIXER6,
+ WM8990_RLBROVOL_SHIFT, WM8990_RLBROVOL_MASK, 1, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX AINRMUX Bypass Volume", WM8990_OUTPUT_MIXER6,
+ WM8990_RRBROVOL_SHIFT, WM8990_RRBROVOL_MASK, 1, out_mix_tlv),
+
+/* LOUT */
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("LOUT Volume", WM8990_LEFT_OUTPUT_VOLUME,
+ WM8990_LOUTVOL_SHIFT, WM8990_LOUTVOL_MASK, 0, out_pga_tlv),
+SOC_SINGLE("LOUT ZC", WM8990_LEFT_OUTPUT_VOLUME, WM8990_LOZC_BIT, 1, 0),
+
+/* ROUT */
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("ROUT Volume", WM8990_RIGHT_OUTPUT_VOLUME,
+ WM8990_ROUTVOL_SHIFT, WM8990_ROUTVOL_MASK, 0, out_pga_tlv),
+SOC_SINGLE("ROUT ZC", WM8990_RIGHT_OUTPUT_VOLUME, WM8990_ROZC_BIT, 1, 0),
+
+/* LOPGA */
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("LOPGA Volume", WM8990_LEFT_OPGA_VOLUME,
+ WM8990_LOPGAVOL_SHIFT, WM8990_LOPGAVOL_MASK, 0, out_pga_tlv),
+SOC_SINGLE("LOPGA ZC Switch", WM8990_LEFT_OPGA_VOLUME,
+ WM8990_LOPGAZC_BIT, 1, 0),
+
+/* ROPGA */
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("ROPGA Volume", WM8990_RIGHT_OPGA_VOLUME,
+ WM8990_ROPGAVOL_SHIFT, WM8990_ROPGAVOL_MASK, 0, out_pga_tlv),
+SOC_SINGLE("ROPGA ZC Switch", WM8990_RIGHT_OPGA_VOLUME,
+ WM8990_ROPGAZC_BIT, 1, 0),
+
+SOC_SINGLE("LON Mute Switch", WM8990_LINE_OUTPUTS_VOLUME,
+ WM8990_LONMUTE_BIT, 1, 0),
+SOC_SINGLE("LOP Mute Switch", WM8990_LINE_OUTPUTS_VOLUME,
+ WM8990_LOPMUTE_BIT, 1, 0),
+SOC_SINGLE("LOP Attenuation Switch", WM8990_LINE_OUTPUTS_VOLUME,
+ WM8990_LOATTN_BIT, 1, 0),
+SOC_SINGLE("RON Mute Switch", WM8990_LINE_OUTPUTS_VOLUME,
+ WM8990_RONMUTE_BIT, 1, 0),
+SOC_SINGLE("ROP Mute Switch", WM8990_LINE_OUTPUTS_VOLUME,
+ WM8990_ROPMUTE_BIT, 1, 0),
+SOC_SINGLE("ROP Attenuation Switch", WM8990_LINE_OUTPUTS_VOLUME,
+ WM8990_ROATTN_BIT, 1, 0),
+
+SOC_SINGLE("OUT3 Mute Switch", WM8990_OUT3_4_VOLUME,
+ WM8990_OUT3MUTE_BIT, 1, 0),
+SOC_SINGLE("OUT3 Attenuation Switch", WM8990_OUT3_4_VOLUME,
+ WM8990_OUT3ATTN_BIT, 1, 0),
+
+SOC_SINGLE("OUT4 Mute Switch", WM8990_OUT3_4_VOLUME,
+ WM8990_OUT4MUTE_BIT, 1, 0),
+SOC_SINGLE("OUT4 Attenuation Switch", WM8990_OUT3_4_VOLUME,
+ WM8990_OUT4ATTN_BIT, 1, 0),
+
+SOC_SINGLE("Speaker Mode Switch", WM8990_CLASSD1,
+ WM8990_CDMODE_BIT, 1, 0),
+
+SOC_SINGLE("Speaker Output Attenuation Volume", WM8990_SPEAKER_VOLUME,
+ WM8990_SPKVOL_SHIFT, WM8990_SPKVOL_MASK, 0),
+SOC_SINGLE("Speaker DC Boost Volume", WM8990_CLASSD3,
+ WM8990_DCGAIN_SHIFT, WM8990_DCGAIN_MASK, 0),
+SOC_SINGLE("Speaker AC Boost Volume", WM8990_CLASSD3,
+ WM8990_ACGAIN_SHIFT, WM8990_ACGAIN_MASK, 0),
+
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("Left DAC Digital Volume",
+ WM8990_LEFT_DAC_DIGITAL_VOLUME,
+ WM8990_DACL_VOL_SHIFT,
+ WM8990_DACL_VOL_MASK,
+ 0,
+ out_dac_tlv),
+
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("Right DAC Digital Volume",
+ WM8990_RIGHT_DAC_DIGITAL_VOLUME,
+ WM8990_DACR_VOL_SHIFT,
+ WM8990_DACR_VOL_MASK,
+ 0,
+ out_dac_tlv),
+
+SOC_ENUM("Left Digital Sidetone", wm8990_left_digital_sidetone_enum),
+SOC_ENUM("Right Digital Sidetone", wm8990_right_digital_sidetone_enum),
+
+SOC_SINGLE_TLV("Left Digital Sidetone Volume", WM8990_DIGITAL_SIDE_TONE,
+ WM8990_ADCL_DAC_SVOL_SHIFT, WM8990_ADCL_DAC_SVOL_MASK, 0,
+ out_sidetone_tlv),
+SOC_SINGLE_TLV("Right Digital Sidetone Volume", WM8990_DIGITAL_SIDE_TONE,
+ WM8990_ADCR_DAC_SVOL_SHIFT, WM8990_ADCR_DAC_SVOL_MASK, 0,
+ out_sidetone_tlv),
+
+SOC_SINGLE("ADC Digital High Pass Filter Switch", WM8990_ADC_CTRL,
+ WM8990_ADC_HPF_ENA_BIT, 1, 0),
+
+SOC_ENUM("ADC HPF Mode", wm8990_right_adcmode_enum),
+
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("Left ADC Digital Volume",
+ WM8990_LEFT_ADC_DIGITAL_VOLUME,
+ WM8990_ADCL_VOL_SHIFT,
+ WM8990_ADCL_VOL_MASK,
+ 0,
+ in_adc_tlv),
+
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("Right ADC Digital Volume",
+ WM8990_RIGHT_ADC_DIGITAL_VOLUME,
+ WM8990_ADCR_VOL_SHIFT,
+ WM8990_ADCR_VOL_MASK,
+ 0,
+ in_adc_tlv),
+
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("LIN12 Volume",
+ WM8990_LEFT_LINE_INPUT_1_2_VOLUME,
+ WM8990_LIN12VOL_SHIFT,
+ WM8990_LIN12VOL_MASK,
+ 0,
+ in_pga_tlv),
+
+SOC_SINGLE("LIN12 ZC Switch", WM8990_LEFT_LINE_INPUT_1_2_VOLUME,
+ WM8990_LI12ZC_BIT, 1, 0),
+
+SOC_SINGLE("LIN12 Mute Switch", WM8990_LEFT_LINE_INPUT_1_2_VOLUME,
+ WM8990_LI12MUTE_BIT, 1, 0),
+
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("LIN34 Volume",
+ WM8990_LEFT_LINE_INPUT_3_4_VOLUME,
+ WM8990_LIN34VOL_SHIFT,
+ WM8990_LIN34VOL_MASK,
+ 0,
+ in_pga_tlv),
+
+SOC_SINGLE("LIN34 ZC Switch", WM8990_LEFT_LINE_INPUT_3_4_VOLUME,
+ WM8990_LI34ZC_BIT, 1, 0),
+
+SOC_SINGLE("LIN34 Mute Switch", WM8990_LEFT_LINE_INPUT_3_4_VOLUME,
+ WM8990_LI34MUTE_BIT, 1, 0),
+
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("RIN12 Volume",
+ WM8990_RIGHT_LINE_INPUT_1_2_VOLUME,
+ WM8990_RIN12VOL_SHIFT,
+ WM8990_RIN12VOL_MASK,
+ 0,
+ in_pga_tlv),
+
+SOC_SINGLE("RIN12 ZC Switch", WM8990_RIGHT_LINE_INPUT_1_2_VOLUME,
+ WM8990_RI12ZC_BIT, 1, 0),
+
+SOC_SINGLE("RIN12 Mute Switch", WM8990_RIGHT_LINE_INPUT_1_2_VOLUME,
+ WM8990_RI12MUTE_BIT, 1, 0),
+
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("RIN34 Volume",
+ WM8990_RIGHT_LINE_INPUT_3_4_VOLUME,
+ WM8990_RIN34VOL_SHIFT,
+ WM8990_RIN34VOL_MASK,
+ 0,
+ in_pga_tlv),
+
+SOC_SINGLE("RIN34 ZC Switch", WM8990_RIGHT_LINE_INPUT_3_4_VOLUME,
+ WM8990_RI34ZC_BIT, 1, 0),
+
+SOC_SINGLE("RIN34 Mute Switch", WM8990_RIGHT_LINE_INPUT_3_4_VOLUME,
+ WM8990_RI34MUTE_BIT, 1, 0),
+
+};
+
+/* add non dapm controls */
+static int wm8990_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(wm8990_snd_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm8990_snd_controls[i], codec,
+ NULL));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+/*
+ * _DAPM_ Controls
+ */
+
+static int inmixer_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ u16 reg, fakepower;
+
+ reg = wm8990_read_reg_cache(w->codec, WM8990_POWER_MANAGEMENT_2);
+ fakepower = wm8990_read_reg_cache(w->codec, WM8990_INTDRIVBITS);
+
+ if (fakepower & ((1 << WM8990_INMIXL_PWR_BIT) |
+ (1 << WM8990_AINLMUX_PWR_BIT))) {
+ reg |= WM8990_AINL_ENA;
+ } else {
+ reg &= ~WM8990_AINL_ENA;
+ }
+
+ if (fakepower & ((1 << WM8990_INMIXR_PWR_BIT) |
+ (1 << WM8990_AINRMUX_PWR_BIT))) {
+ reg |= WM8990_AINR_ENA;
+ } else {
+ reg &= ~WM8990_AINL_ENA;
+ }
+ wm8990_write(w->codec, WM8990_POWER_MANAGEMENT_2, reg);
+
+ return 0;
+}
+
+static int outmixer_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ u32 reg_shift = kcontrol->private_value & 0xfff;
+ int ret = 0;
+ u16 reg;
+
+ switch (reg_shift) {
+ case WM8990_SPEAKER_MIXER | (WM8990_LDSPK_BIT << 8) :
+ reg = wm8990_read_reg_cache(w->codec, WM8990_OUTPUT_MIXER1);
+ if (reg & WM8990_LDLO) {
+ printk(KERN_WARNING
+ "Cannot set as Output Mixer 1 LDLO Set\n");
+ ret = -1;
+ }
+ break;
+ case WM8990_SPEAKER_MIXER | (WM8990_RDSPK_BIT << 8):
+ reg = wm8990_read_reg_cache(w->codec, WM8990_OUTPUT_MIXER2);
+ if (reg & WM8990_RDRO) {
+ printk(KERN_WARNING
+ "Cannot set as Output Mixer 2 RDRO Set\n");
+ ret = -1;
+ }
+ break;
+ case WM8990_OUTPUT_MIXER1 | (WM8990_LDLO_BIT << 8):
+ reg = wm8990_read_reg_cache(w->codec, WM8990_SPEAKER_MIXER);
+ if (reg & WM8990_LDSPK) {
+ printk(KERN_WARNING
+ "Cannot set as Speaker Mixer LDSPK Set\n");
+ ret = -1;
+ }
+ break;
+ case WM8990_OUTPUT_MIXER2 | (WM8990_RDRO_BIT << 8):
+ reg = wm8990_read_reg_cache(w->codec, WM8990_SPEAKER_MIXER);
+ if (reg & WM8990_RDSPK) {
+ printk(KERN_WARNING
+ "Cannot set as Speaker Mixer RDSPK Set\n");
+ ret = -1;
+ }
+ break;
+ }
+
+ return ret;
+}
+
+/* INMIX dB values */
+static const unsigned int in_mix_tlv[] = {
+ TLV_DB_RANGE_HEAD(1),
+ 0, 7, TLV_DB_LINEAR_ITEM(-1200, 600),
+};
+
+/* Left In PGA Connections */
+static const struct snd_kcontrol_new wm8990_dapm_lin12_pga_controls[] = {
+SOC_DAPM_SINGLE("LIN1 Switch", WM8990_INPUT_MIXER2, WM8990_LMN1_BIT, 1, 0),
+SOC_DAPM_SINGLE("LIN2 Switch", WM8990_INPUT_MIXER2, WM8990_LMP2_BIT, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8990_dapm_lin34_pga_controls[] = {
+SOC_DAPM_SINGLE("LIN3 Switch", WM8990_INPUT_MIXER2, WM8990_LMN3_BIT, 1, 0),
+SOC_DAPM_SINGLE("LIN4 Switch", WM8990_INPUT_MIXER2, WM8990_LMP4_BIT, 1, 0),
+};
+
+/* Right In PGA Connections */
+static const struct snd_kcontrol_new wm8990_dapm_rin12_pga_controls[] = {
+SOC_DAPM_SINGLE("RIN1 Switch", WM8990_INPUT_MIXER2, WM8990_RMN1_BIT, 1, 0),
+SOC_DAPM_SINGLE("RIN2 Switch", WM8990_INPUT_MIXER2, WM8990_RMP2_BIT, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8990_dapm_rin34_pga_controls[] = {
+SOC_DAPM_SINGLE("RIN3 Switch", WM8990_INPUT_MIXER2, WM8990_RMN3_BIT, 1, 0),
+SOC_DAPM_SINGLE("RIN4 Switch", WM8990_INPUT_MIXER2, WM8990_RMP4_BIT, 1, 0),
+};
+
+/* INMIXL */
+static const struct snd_kcontrol_new wm8990_dapm_inmixl_controls[] = {
+SOC_DAPM_SINGLE_TLV("Record Left Volume", WM8990_INPUT_MIXER3,
+ WM8990_LDBVOL_SHIFT, WM8990_LDBVOL_MASK, 0, in_mix_tlv),
+SOC_DAPM_SINGLE_TLV("LIN2 Volume", WM8990_INPUT_MIXER5, WM8990_LI2BVOL_SHIFT,
+ 7, 0, in_mix_tlv),
+SOC_DAPM_SINGLE("LINPGA12 Switch", WM8990_INPUT_MIXER3, WM8990_L12MNB_BIT,
+ 1, 0),
+SOC_DAPM_SINGLE("LINPGA34 Switch", WM8990_INPUT_MIXER3, WM8990_L34MNB_BIT,
+ 1, 0),
+};
+
+/* INMIXR */
+static const struct snd_kcontrol_new wm8990_dapm_inmixr_controls[] = {
+SOC_DAPM_SINGLE_TLV("Record Right Volume", WM8990_INPUT_MIXER4,
+ WM8990_RDBVOL_SHIFT, WM8990_RDBVOL_MASK, 0, in_mix_tlv),
+SOC_DAPM_SINGLE_TLV("RIN2 Volume", WM8990_INPUT_MIXER6, WM8990_RI2BVOL_SHIFT,
+ 7, 0, in_mix_tlv),
+SOC_DAPM_SINGLE("RINPGA12 Switch", WM8990_INPUT_MIXER3, WM8990_L12MNB_BIT,
+ 1, 0),
+SOC_DAPM_SINGLE("RINPGA34 Switch", WM8990_INPUT_MIXER3, WM8990_L34MNB_BIT,
+ 1, 0),
+};
+
+/* AINLMUX */
+static const char *wm8990_ainlmux[] =
+ {"INMIXL Mix", "RXVOICE Mix", "DIFFINL Mix"};
+
+static const struct soc_enum wm8990_ainlmux_enum =
+SOC_ENUM_SINGLE(WM8990_INPUT_MIXER1, WM8990_AINLMODE_SHIFT,
+ ARRAY_SIZE(wm8990_ainlmux), wm8990_ainlmux);
+
+static const struct snd_kcontrol_new wm8990_dapm_ainlmux_controls =
+SOC_DAPM_ENUM("Route", wm8990_ainlmux_enum);
+
+/* DIFFINL */
+
+/* AINRMUX */
+static const char *wm8990_ainrmux[] =
+ {"INMIXR Mix", "RXVOICE Mix", "DIFFINR Mix"};
+
+static const struct soc_enum wm8990_ainrmux_enum =
+SOC_ENUM_SINGLE(WM8990_INPUT_MIXER1, WM8990_AINRMODE_SHIFT,
+ ARRAY_SIZE(wm8990_ainrmux), wm8990_ainrmux);
+
+static const struct snd_kcontrol_new wm8990_dapm_ainrmux_controls =
+SOC_DAPM_ENUM("Route", wm8990_ainrmux_enum);
+
+/* RXVOICE */
+static const struct snd_kcontrol_new wm8990_dapm_rxvoice_controls[] = {
+SOC_DAPM_SINGLE_TLV("LIN4/RXN", WM8990_INPUT_MIXER5, WM8990_LR4BVOL_SHIFT,
+ WM8990_LR4BVOL_MASK, 0, in_mix_tlv),
+SOC_DAPM_SINGLE_TLV("RIN4/RXP", WM8990_INPUT_MIXER6, WM8990_RL4BVOL_SHIFT,
+ WM8990_RL4BVOL_MASK, 0, in_mix_tlv),
+};
+
+/* LOMIX */
+static const struct snd_kcontrol_new wm8990_dapm_lomix_controls[] = {
+SOC_DAPM_SINGLE("LOMIX Right ADC Bypass Switch", WM8990_OUTPUT_MIXER1,
+ WM8990_LRBLO_BIT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX Left ADC Bypass Switch", WM8990_OUTPUT_MIXER1,
+ WM8990_LLBLO_BIT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX RIN3 Bypass Switch", WM8990_OUTPUT_MIXER1,
+ WM8990_LRI3LO_BIT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX LIN3 Bypass Switch", WM8990_OUTPUT_MIXER1,
+ WM8990_LLI3LO_BIT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX RIN12 PGA Bypass Switch", WM8990_OUTPUT_MIXER1,
+ WM8990_LR12LO_BIT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX LIN12 PGA Bypass Switch", WM8990_OUTPUT_MIXER1,
+ WM8990_LL12LO_BIT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX Left DAC Switch", WM8990_OUTPUT_MIXER1,
+ WM8990_LDLO_BIT, 1, 0),
+};
+
+/* ROMIX */
+static const struct snd_kcontrol_new wm8990_dapm_romix_controls[] = {
+SOC_DAPM_SINGLE("ROMIX Left ADC Bypass Switch", WM8990_OUTPUT_MIXER2,
+ WM8990_RLBRO_BIT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX Right ADC Bypass Switch", WM8990_OUTPUT_MIXER2,
+ WM8990_RRBRO_BIT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX LIN3 Bypass Switch", WM8990_OUTPUT_MIXER2,
+ WM8990_RLI3RO_BIT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX RIN3 Bypass Switch", WM8990_OUTPUT_MIXER2,
+ WM8990_RRI3RO_BIT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX LIN12 PGA Bypass Switch", WM8990_OUTPUT_MIXER2,
+ WM8990_RL12RO_BIT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX RIN12 PGA Bypass Switch", WM8990_OUTPUT_MIXER2,
+ WM8990_RR12RO_BIT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX Right DAC Switch", WM8990_OUTPUT_MIXER2,
+ WM8990_RDRO_BIT, 1, 0),
+};
+
+/* LONMIX */
+static const struct snd_kcontrol_new wm8990_dapm_lonmix_controls[] = {
+SOC_DAPM_SINGLE("LONMIX Left Mixer PGA Switch", WM8990_LINE_MIXER1,
+ WM8990_LLOPGALON_BIT, 1, 0),
+SOC_DAPM_SINGLE("LONMIX Right Mixer PGA Switch", WM8990_LINE_MIXER1,
+ WM8990_LROPGALON_BIT, 1, 0),
+SOC_DAPM_SINGLE("LONMIX Inverted LOP Switch", WM8990_LINE_MIXER1,
+ WM8990_LOPLON_BIT, 1, 0),
+};
+
+/* LOPMIX */
+static const struct snd_kcontrol_new wm8990_dapm_lopmix_controls[] = {
+SOC_DAPM_SINGLE("LOPMIX Right Mic Bypass Switch", WM8990_LINE_MIXER1,
+ WM8990_LR12LOP_BIT, 1, 0),
+SOC_DAPM_SINGLE("LOPMIX Left Mic Bypass Switch", WM8990_LINE_MIXER1,
+ WM8990_LL12LOP_BIT, 1, 0),
+SOC_DAPM_SINGLE("LOPMIX Left Mixer PGA Switch", WM8990_LINE_MIXER1,
+ WM8990_LLOPGALOP_BIT, 1, 0),
+};
+
+/* RONMIX */
+static const struct snd_kcontrol_new wm8990_dapm_ronmix_controls[] = {
+SOC_DAPM_SINGLE("RONMIX Right Mixer PGA Switch", WM8990_LINE_MIXER2,
+ WM8990_RROPGARON_BIT, 1, 0),
+SOC_DAPM_SINGLE("RONMIX Left Mixer PGA Switch", WM8990_LINE_MIXER2,
+ WM8990_RLOPGARON_BIT, 1, 0),
+SOC_DAPM_SINGLE("RONMIX Inverted ROP Switch", WM8990_LINE_MIXER2,
+ WM8990_ROPRON_BIT, 1, 0),
+};
+
+/* ROPMIX */
+static const struct snd_kcontrol_new wm8990_dapm_ropmix_controls[] = {
+SOC_DAPM_SINGLE("ROPMIX Left Mic Bypass Switch", WM8990_LINE_MIXER2,
+ WM8990_RL12ROP_BIT, 1, 0),
+SOC_DAPM_SINGLE("ROPMIX Right Mic Bypass Switch", WM8990_LINE_MIXER2,
+ WM8990_RR12ROP_BIT, 1, 0),
+SOC_DAPM_SINGLE("ROPMIX Right Mixer PGA Switch", WM8990_LINE_MIXER2,
+ WM8990_RROPGAROP_BIT, 1, 0),
+};
+
+/* OUT3MIX */
+static const struct snd_kcontrol_new wm8990_dapm_out3mix_controls[] = {
+SOC_DAPM_SINGLE("OUT3MIX LIN4/RXP Bypass Switch", WM8990_OUT3_4_MIXER,
+ WM8990_LI4O3_BIT, 1, 0),
+SOC_DAPM_SINGLE("OUT3MIX Left Out PGA Switch", WM8990_OUT3_4_MIXER,
+ WM8990_LPGAO3_BIT, 1, 0),
+};
+
+/* OUT4MIX */
+static const struct snd_kcontrol_new wm8990_dapm_out4mix_controls[] = {
+SOC_DAPM_SINGLE("OUT4MIX Right Out PGA Switch", WM8990_OUT3_4_MIXER,
+ WM8990_RPGAO4_BIT, 1, 0),
+SOC_DAPM_SINGLE("OUT4MIX RIN4/RXP Bypass Switch", WM8990_OUT3_4_MIXER,
+ WM8990_RI4O4_BIT, 1, 0),
+};
+
+/* SPKMIX */
+static const struct snd_kcontrol_new wm8990_dapm_spkmix_controls[] = {
+SOC_DAPM_SINGLE("SPKMIX LIN2 Bypass Switch", WM8990_SPEAKER_MIXER,
+ WM8990_LI2SPK_BIT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX LADC Bypass Switch", WM8990_SPEAKER_MIXER,
+ WM8990_LB2SPK_BIT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX Left Mixer PGA Switch", WM8990_SPEAKER_MIXER,
+ WM8990_LOPGASPK_BIT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX Left DAC Switch", WM8990_SPEAKER_MIXER,
+ WM8990_LDSPK_BIT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX Right DAC Switch", WM8990_SPEAKER_MIXER,
+ WM8990_RDSPK_BIT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX Right Mixer PGA Switch", WM8990_SPEAKER_MIXER,
+ WM8990_ROPGASPK_BIT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX RADC Bypass Switch", WM8990_SPEAKER_MIXER,
+ WM8990_RL12ROP_BIT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX RIN2 Bypass Switch", WM8990_SPEAKER_MIXER,
+ WM8990_RI2SPK_BIT, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget wm8990_dapm_widgets[] = {
+/* Input Side */
+/* Input Lines */
+SND_SOC_DAPM_INPUT("LIN1"),
+SND_SOC_DAPM_INPUT("LIN2"),
+SND_SOC_DAPM_INPUT("LIN3"),
+SND_SOC_DAPM_INPUT("LIN4/RXN"),
+SND_SOC_DAPM_INPUT("RIN3"),
+SND_SOC_DAPM_INPUT("RIN4/RXP"),
+SND_SOC_DAPM_INPUT("RIN1"),
+SND_SOC_DAPM_INPUT("RIN2"),
+SND_SOC_DAPM_INPUT("Internal ADC Source"),
+
+/* DACs */
+SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8990_POWER_MANAGEMENT_2,
+ WM8990_ADCL_ENA_BIT, 0),
+SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8990_POWER_MANAGEMENT_2,
+ WM8990_ADCR_ENA_BIT, 0),
+
+/* Input PGAs */
+SND_SOC_DAPM_MIXER("LIN12 PGA", WM8990_POWER_MANAGEMENT_2, WM8990_LIN12_ENA_BIT,
+ 0, &wm8990_dapm_lin12_pga_controls[0],
+ ARRAY_SIZE(wm8990_dapm_lin12_pga_controls)),
+SND_SOC_DAPM_MIXER("LIN34 PGA", WM8990_POWER_MANAGEMENT_2, WM8990_LIN34_ENA_BIT,
+ 0, &wm8990_dapm_lin34_pga_controls[0],
+ ARRAY_SIZE(wm8990_dapm_lin34_pga_controls)),
+SND_SOC_DAPM_MIXER("RIN12 PGA", WM8990_POWER_MANAGEMENT_2, WM8990_RIN12_ENA_BIT,
+ 0, &wm8990_dapm_rin12_pga_controls[0],
+ ARRAY_SIZE(wm8990_dapm_rin12_pga_controls)),
+SND_SOC_DAPM_MIXER("RIN34 PGA", WM8990_POWER_MANAGEMENT_2, WM8990_RIN34_ENA_BIT,
+ 0, &wm8990_dapm_rin34_pga_controls[0],
+ ARRAY_SIZE(wm8990_dapm_rin34_pga_controls)),
+
+/* INMIXL */
+SND_SOC_DAPM_MIXER_E("INMIXL", WM8990_INTDRIVBITS, WM8990_INMIXL_PWR_BIT, 0,
+ &wm8990_dapm_inmixl_controls[0],
+ ARRAY_SIZE(wm8990_dapm_inmixl_controls),
+ inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+/* AINLMUX */
+SND_SOC_DAPM_MUX_E("AILNMUX", WM8990_INTDRIVBITS, WM8990_AINLMUX_PWR_BIT, 0,
+ &wm8990_dapm_ainlmux_controls, inmixer_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+/* INMIXR */
+SND_SOC_DAPM_MIXER_E("INMIXR", WM8990_INTDRIVBITS, WM8990_INMIXR_PWR_BIT, 0,
+ &wm8990_dapm_inmixr_controls[0],
+ ARRAY_SIZE(wm8990_dapm_inmixr_controls),
+ inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+/* AINRMUX */
+SND_SOC_DAPM_MUX_E("AIRNMUX", WM8990_INTDRIVBITS, WM8990_AINRMUX_PWR_BIT, 0,
+ &wm8990_dapm_ainrmux_controls, inmixer_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+/* Output Side */
+/* DACs */
+SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8990_POWER_MANAGEMENT_3,
+ WM8990_DACL_ENA_BIT, 0),
+SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8990_POWER_MANAGEMENT_3,
+ WM8990_DACR_ENA_BIT, 0),
+
+/* LOMIX */
+SND_SOC_DAPM_MIXER_E("LOMIX", WM8990_POWER_MANAGEMENT_3, WM8990_LOMIX_ENA_BIT,
+ 0, &wm8990_dapm_lomix_controls[0],
+ ARRAY_SIZE(wm8990_dapm_lomix_controls),
+ outmixer_event, SND_SOC_DAPM_PRE_REG),
+
+/* LONMIX */
+SND_SOC_DAPM_MIXER("LONMIX", WM8990_POWER_MANAGEMENT_3, WM8990_LON_ENA_BIT, 0,
+ &wm8990_dapm_lonmix_controls[0],
+ ARRAY_SIZE(wm8990_dapm_lonmix_controls)),
+
+/* LOPMIX */
+SND_SOC_DAPM_MIXER("LOPMIX", WM8990_POWER_MANAGEMENT_3, WM8990_LOP_ENA_BIT, 0,
+ &wm8990_dapm_lopmix_controls[0],
+ ARRAY_SIZE(wm8990_dapm_lopmix_controls)),
+
+/* OUT3MIX */
+SND_SOC_DAPM_MIXER("OUT3MIX", WM8990_POWER_MANAGEMENT_1, WM8990_OUT3_ENA_BIT, 0,
+ &wm8990_dapm_out3mix_controls[0],
+ ARRAY_SIZE(wm8990_dapm_out3mix_controls)),
+
+/* SPKMIX */
+SND_SOC_DAPM_MIXER_E("SPKMIX", WM8990_POWER_MANAGEMENT_1, WM8990_SPK_ENA_BIT, 0,
+ &wm8990_dapm_spkmix_controls[0],
+ ARRAY_SIZE(wm8990_dapm_spkmix_controls), outmixer_event,
+ SND_SOC_DAPM_PRE_REG),
+
+/* OUT4MIX */
+SND_SOC_DAPM_MIXER("OUT4MIX", WM8990_POWER_MANAGEMENT_1, WM8990_OUT4_ENA_BIT, 0,
+ &wm8990_dapm_out4mix_controls[0],
+ ARRAY_SIZE(wm8990_dapm_out4mix_controls)),
+
+/* ROPMIX */
+SND_SOC_DAPM_MIXER("ROPMIX", WM8990_POWER_MANAGEMENT_3, WM8990_ROP_ENA_BIT, 0,
+ &wm8990_dapm_ropmix_controls[0],
+ ARRAY_SIZE(wm8990_dapm_ropmix_controls)),
+
+/* RONMIX */
+SND_SOC_DAPM_MIXER("RONMIX", WM8990_POWER_MANAGEMENT_3, WM8990_RON_ENA_BIT, 0,
+ &wm8990_dapm_ronmix_controls[0],
+ ARRAY_SIZE(wm8990_dapm_ronmix_controls)),
+
+/* ROMIX */
+SND_SOC_DAPM_MIXER_E("ROMIX", WM8990_POWER_MANAGEMENT_3, WM8990_ROMIX_ENA_BIT,
+ 0, &wm8990_dapm_romix_controls[0],
+ ARRAY_SIZE(wm8990_dapm_romix_controls),
+ outmixer_event, SND_SOC_DAPM_PRE_REG),
+
+/* LOUT PGA */
+SND_SOC_DAPM_PGA("LOUT PGA", WM8990_POWER_MANAGEMENT_1, WM8990_LOUT_ENA_BIT, 0,
+ NULL, 0),
+
+/* ROUT PGA */
+SND_SOC_DAPM_PGA("ROUT PGA", WM8990_POWER_MANAGEMENT_1, WM8990_ROUT_ENA_BIT, 0,
+ NULL, 0),
+
+/* LOPGA */
+SND_SOC_DAPM_PGA("LOPGA", WM8990_POWER_MANAGEMENT_3, WM8990_LOPGA_ENA_BIT, 0,
+ NULL, 0),
+
+/* ROPGA */
+SND_SOC_DAPM_PGA("ROPGA", WM8990_POWER_MANAGEMENT_3, WM8990_ROPGA_ENA_BIT, 0,
+ NULL, 0),
+
+/* MICBIAS */
+SND_SOC_DAPM_MICBIAS("MICBIAS", WM8990_POWER_MANAGEMENT_1,
+ WM8990_MICBIAS_ENA_BIT, 0),
+
+SND_SOC_DAPM_OUTPUT("LON"),
+SND_SOC_DAPM_OUTPUT("LOP"),
+SND_SOC_DAPM_OUTPUT("OUT3"),
+SND_SOC_DAPM_OUTPUT("LOUT"),
+SND_SOC_DAPM_OUTPUT("SPKN"),
+SND_SOC_DAPM_OUTPUT("SPKP"),
+SND_SOC_DAPM_OUTPUT("ROUT"),
+SND_SOC_DAPM_OUTPUT("OUT4"),
+SND_SOC_DAPM_OUTPUT("ROP"),
+SND_SOC_DAPM_OUTPUT("RON"),
+
+SND_SOC_DAPM_OUTPUT("Internal DAC Sink"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ /* Make DACs turn on when playing even if not mixed into any outputs */
+ {"Internal DAC Sink", NULL, "Left DAC"},
+ {"Internal DAC Sink", NULL, "Right DAC"},
+
+ /* Make ADCs turn on when recording even if not mixed from any inputs */
+ {"Left ADC", NULL, "Internal ADC Source"},
+ {"Right ADC", NULL, "Internal ADC Source"},
+
+ /* Input Side */
+ /* LIN12 PGA */
+ {"LIN12 PGA", "LIN1 Switch", "LIN1"},
+ {"LIN12 PGA", "LIN2 Switch", "LIN2"},
+ /* LIN34 PGA */
+ {"LIN34 PGA", "LIN3 Switch", "LIN3"},
+ {"LIN34 PGA", "LIN4 Switch", "LIN4"},
+ /* INMIXL */
+ {"INMIXL", "Record Left Volume", "LOMIX"},
+ {"INMIXL", "LIN2 Volume", "LIN2"},
+ {"INMIXL", "LINPGA12 Switch", "LIN12 PGA"},
+ {"INMIXL", "LINPGA34 Switch", "LIN34 PGA"},
+ /* AILNMUX */
+ {"AILNMUX", "INMIXL Mix", "INMIXL"},
+ {"AILNMUX", "DIFFINL Mix", "LIN12PGA"},
+ {"AILNMUX", "DIFFINL Mix", "LIN34PGA"},
+ {"AILNMUX", "RXVOICE Mix", "LIN4/RXN"},
+ {"AILNMUX", "RXVOICE Mix", "RIN4/RXP"},
+ /* ADC */
+ {"Left ADC", NULL, "AILNMUX"},
+
+ /* RIN12 PGA */
+ {"RIN12 PGA", "RIN1 Switch", "RIN1"},
+ {"RIN12 PGA", "RIN2 Switch", "RIN2"},
+ /* RIN34 PGA */
+ {"RIN34 PGA", "RIN3 Switch", "RIN3"},
+ {"RIN34 PGA", "RIN4 Switch", "RIN4"},
+ /* INMIXL */
+ {"INMIXR", "Record Right Volume", "ROMIX"},
+ {"INMIXR", "RIN2 Volume", "RIN2"},
+ {"INMIXR", "RINPGA12 Switch", "RIN12 PGA"},
+ {"INMIXR", "RINPGA34 Switch", "RIN34 PGA"},
+ /* AIRNMUX */
+ {"AIRNMUX", "INMIXR Mix", "INMIXR"},
+ {"AIRNMUX", "DIFFINR Mix", "RIN12PGA"},
+ {"AIRNMUX", "DIFFINR Mix", "RIN34PGA"},
+ {"AIRNMUX", "RXVOICE Mix", "RIN4/RXN"},
+ {"AIRNMUX", "RXVOICE Mix", "RIN4/RXP"},
+ /* ADC */
+ {"Right ADC", NULL, "AIRNMUX"},
+
+ /* LOMIX */
+ {"LOMIX", "LOMIX RIN3 Bypass Switch", "RIN3"},
+ {"LOMIX", "LOMIX LIN3 Bypass Switch", "LIN3"},
+ {"LOMIX", "LOMIX LIN12 PGA Bypass Switch", "LIN12 PGA"},
+ {"LOMIX", "LOMIX RIN12 PGA Bypass Switch", "RIN12 PGA"},
+ {"LOMIX", "LOMIX Right ADC Bypass Switch", "AINRMUX"},
+ {"LOMIX", "LOMIX Left ADC Bypass Switch", "AINLMUX"},
+ {"LOMIX", "LOMIX Left DAC Switch", "Left DAC"},
+
+ /* ROMIX */
+ {"ROMIX", "ROMIX RIN3 Bypass Switch", "RIN3"},
+ {"ROMIX", "ROMIX LIN3 Bypass Switch", "LIN3"},
+ {"ROMIX", "ROMIX LIN12 PGA Bypass Switch", "LIN12 PGA"},
+ {"ROMIX", "ROMIX RIN12 PGA Bypass Switch", "RIN12 PGA"},
+ {"ROMIX", "ROMIX Right ADC Bypass Switch", "AINRMUX"},
+ {"ROMIX", "ROMIX Left ADC Bypass Switch", "AINLMUX"},
+ {"ROMIX", "ROMIX Right DAC Switch", "Right DAC"},
+
+ /* SPKMIX */
+ {"SPKMIX", "SPKMIX LIN2 Bypass Switch", "LIN2"},
+ {"SPKMIX", "SPKMIX RIN2 Bypass Switch", "RIN2"},
+ {"SPKMIX", "SPKMIX LADC Bypass Switch", "AINLMUX"},
+ {"SPKMIX", "SPKMIX RADC Bypass Switch", "AINRMUX"},
+ {"SPKMIX", "SPKMIX Left Mixer PGA Switch", "LOPGA"},
+ {"SPKMIX", "SPKMIX Right Mixer PGA Switch", "ROPGA"},
+ {"SPKMIX", "SPKMIX Right DAC Switch", "Right DAC"},
+ {"SPKMIX", "SPKMIX Left DAC Switch", "Right DAC"},
+
+ /* LONMIX */
+ {"LONMIX", "LONMIX Left Mixer PGA Switch", "LOPGA"},
+ {"LONMIX", "LONMIX Right Mixer PGA Switch", "ROPGA"},
+ {"LONMIX", "LONMIX Inverted LOP Switch", "LOPMIX"},
+
+ /* LOPMIX */
+ {"LOPMIX", "LOPMIX Right Mic Bypass Switch", "RIN12 PGA"},
+ {"LOPMIX", "LOPMIX Left Mic Bypass Switch", "LIN12 PGA"},
+ {"LOPMIX", "LOPMIX Left Mixer PGA Switch", "LOPGA"},
+
+ /* OUT3MIX */
+ {"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXP"},
+ {"OUT3MIX", "OUT3MIX Left Out PGA Switch", "LOPGA"},
+
+ /* OUT4MIX */
+ {"OUT4MIX", "OUT4MIX Right Out PGA Switch", "ROPGA"},
+ {"OUT4MIX", "OUT4MIX RIN4/RXP Bypass Switch", "RIN4/RXP"},
+
+ /* RONMIX */
+ {"RONMIX", "RONMIX Right Mixer PGA Switch", "ROPGA"},
+ {"RONMIX", "RONMIX Left Mixer PGA Switch", "LOPGA"},
+ {"RONMIX", "RONMIX Inverted ROP Switch", "ROPMIX"},
+
+ /* ROPMIX */
+ {"ROPMIX", "ROPMIX Left Mic Bypass Switch", "LIN12 PGA"},
+ {"ROPMIX", "ROPMIX Right Mic Bypass Switch", "RIN12 PGA"},
+ {"ROPMIX", "ROPMIX Right Mixer PGA Switch", "ROPGA"},
+
+ /* Out Mixer PGAs */
+ {"LOPGA", NULL, "LOMIX"},
+ {"ROPGA", NULL, "ROMIX"},
+
+ {"LOUT PGA", NULL, "LOMIX"},
+ {"ROUT PGA", NULL, "ROMIX"},
+
+ /* Output Pins */
+ {"LON", NULL, "LONMIX"},
+ {"LOP", NULL, "LOPMIX"},
+ {"OUT", NULL, "OUT3MIX"},
+ {"LOUT", NULL, "LOUT PGA"},
+ {"SPKN", NULL, "SPKMIX"},
+ {"ROUT", NULL, "ROUT PGA"},
+ {"OUT4", NULL, "OUT4MIX"},
+ {"ROP", NULL, "ROPMIX"},
+ {"RON", NULL, "RONMIX"},
+};
+
+static int wm8990_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, wm8990_dapm_widgets,
+ ARRAY_SIZE(wm8990_dapm_widgets));
+
+ /* set up the WM8990 audio map */
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+/* PLL divisors */
+struct _pll_div {
+ u32 div2;
+ u32 n;
+ u32 k;
+};
+
+/* The size in bits of the pll divide multiplied by 10
+ * to allow rounding later */
+#define FIXED_PLL_SIZE ((1 << 16) * 10)
+
+static void pll_factors(struct _pll_div *pll_div, unsigned int target,
+ unsigned int source)
+{
+ u64 Kpart;
+ unsigned int K, Ndiv, Nmod;
+
+
+ Ndiv = target / source;
+ if (Ndiv < 6) {
+ source >>= 1;
+ pll_div->div2 = 1;
+ Ndiv = target / source;
+ } else
+ pll_div->div2 = 0;
+
+ if ((Ndiv < 6) || (Ndiv > 12))
+ printk(KERN_WARNING
+ "WM8990 N value outwith recommended range! N = %d\n", Ndiv);
+
+ pll_div->n = Ndiv;
+ Nmod = target % source;
+ Kpart = FIXED_PLL_SIZE * (long long)Nmod;
+
+ do_div(Kpart, source);
+
+ K = Kpart & 0xFFFFFFFF;
+
+ /* Check if we need to round */
+ if ((K % 10) >= 5)
+ K += 5;
+
+ /* Move down to proper range now rounding is done */
+ K /= 10;
+
+ pll_div->k = K;
+}
+
+static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai,
+ int pll_id, unsigned int freq_in, unsigned int freq_out)
+{
+ u16 reg;
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct _pll_div pll_div;
+
+ if (freq_in && freq_out) {
+ pll_factors(&pll_div, freq_out * 4, freq_in);
+
+ /* Turn on PLL */
+ reg = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_2);
+ reg |= WM8990_PLL_ENA;
+ wm8990_write(codec, WM8990_POWER_MANAGEMENT_2, reg);
+
+ /* sysclk comes from PLL */
+ reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_2);
+ wm8990_write(codec, WM8990_CLOCKING_2, reg | WM8990_SYSCLK_SRC);
+
+ /* set up N , fractional mode and pre-divisor if neccessary */
+ wm8990_write(codec, WM8990_PLL1, pll_div.n | WM8990_SDM |
+ (pll_div.div2?WM8990_PRESCALE:0));
+ wm8990_write(codec, WM8990_PLL2, (u8)(pll_div.k>>8));
+ wm8990_write(codec, WM8990_PLL3, (u8)(pll_div.k & 0xFF));
+ } else {
+ /* Turn on PLL */
+ reg = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_2);
+ reg &= ~WM8990_PLL_ENA;
+ wm8990_write(codec, WM8990_POWER_MANAGEMENT_2, reg);
+ }
+ return 0;
+}
+
+/*
+ * Clock after PLL and dividers
+ */
+static int wm8990_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 wm8990_priv *wm8990 = codec->private_data;
+
+ wm8990->sysclk = freq;
+ return 0;
+}
+
+/*
+ * Set's ADC and Voice DAC format.
+ */
+static int wm8990_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 audio1, audio3;
+
+ audio1 = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_1);
+ audio3 = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_3);
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ audio3 &= ~WM8990_AIF_MSTR1;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ audio3 |= WM8990_AIF_MSTR1;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ audio1 &= ~WM8990_AIF_FMT_MASK;
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ audio1 |= WM8990_AIF_TMF_I2S;
+ audio1 &= ~WM8990_AIF_LRCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ audio1 |= WM8990_AIF_TMF_RIGHTJ;
+ audio1 &= ~WM8990_AIF_LRCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ audio1 |= WM8990_AIF_TMF_LEFTJ;
+ audio1 &= ~WM8990_AIF_LRCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ audio1 |= WM8990_AIF_TMF_DSP;
+ audio1 &= ~WM8990_AIF_LRCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ audio1 |= WM8990_AIF_TMF_DSP | WM8990_AIF_LRCLK_INV;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ wm8990_write(codec, WM8990_AUDIO_INTERFACE_1, audio1);
+ wm8990_write(codec, WM8990_AUDIO_INTERFACE_3, audio3);
+ return 0;
+}
+
+static int wm8990_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
+ int div_id, int div)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 reg;
+
+ switch (div_id) {
+ case WM8990_MCLK_DIV:
+ reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_2) &
+ ~WM8990_MCLK_DIV_MASK;
+ wm8990_write(codec, WM8990_CLOCKING_2, reg | div);
+ break;
+ case WM8990_DACCLK_DIV:
+ reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_2) &
+ ~WM8990_DAC_CLKDIV_MASK;
+ wm8990_write(codec, WM8990_CLOCKING_2, reg | div);
+ break;
+ case WM8990_ADCCLK_DIV:
+ reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_2) &
+ ~WM8990_ADC_CLKDIV_MASK;
+ wm8990_write(codec, WM8990_CLOCKING_2, reg | div);
+ break;
+ case WM8990_BCLK_DIV:
+ reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_1) &
+ ~WM8990_BCLK_DIV_MASK;
+ wm8990_write(codec, WM8990_CLOCKING_1, reg | div);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+/*
+ * Set PCM DAI bit size and sample rate.
+ */
+static int wm8990_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ u16 audio1 = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_1);
+
+ audio1 &= ~WM8990_AIF_WL_MASK;
+ /* bit size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ audio1 |= WM8990_AIF_WL_20BITS;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ audio1 |= WM8990_AIF_WL_24BITS;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ audio1 |= WM8990_AIF_WL_32BITS;
+ break;
+ }
+
+ wm8990_write(codec, WM8990_AUDIO_INTERFACE_1, audio1);
+ return 0;
+}
+
+static int wm8990_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 val;
+
+ val = wm8990_read_reg_cache(codec, WM8990_DAC_CTRL) & ~WM8990_DAC_MUTE;
+
+ if (mute)
+ wm8990_write(codec, WM8990_DAC_CTRL, val | WM8990_DAC_MUTE);
+ else
+ wm8990_write(codec, WM8990_DAC_CTRL, val);
+
+ return 0;
+}
+
+static int wm8990_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u16 val;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ /* Enable all output discharge bits */
+ wm8990_write(codec, WM8990_ANTIPOP1, WM8990_DIS_LLINE |
+ WM8990_DIS_RLINE | WM8990_DIS_OUT3 |
+ WM8990_DIS_OUT4 | WM8990_DIS_LOUT |
+ WM8990_DIS_ROUT);
+
+ /* Enable POBCTRL, SOFT_ST, VMIDTOG and BUFDCOPEN */
+ wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST |
+ WM8990_BUFDCOPEN | WM8990_POBCTRL |
+ WM8990_VMIDTOG);
+
+ /* Delay to allow output caps to discharge */
+ msleep(msecs_to_jiffies(300));
+
+ /* Disable VMIDTOG */
+ wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST |
+ WM8990_BUFDCOPEN | WM8990_POBCTRL);
+
+ /* disable all output discharge bits */
+ wm8990_write(codec, WM8990_ANTIPOP1, 0);
+
+ /* Enable outputs */
+ wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1b00);
+
+ msleep(msecs_to_jiffies(50));
+
+ /* Enable VMID at 2x50k */
+ wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f02);
+
+ msleep(msecs_to_jiffies(100));
+
+ /* Enable VREF */
+ wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f03);
+
+ msleep(msecs_to_jiffies(600));
+
+ /* Enable BUFIOEN */
+ wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST |
+ WM8990_BUFDCOPEN | WM8990_POBCTRL |
+ WM8990_BUFIOEN);
+
+ /* Disable outputs */
+ wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x3);
+
+ /* disable POBCTRL, SOFT_ST and BUFDCOPEN */
+ wm8990_write(codec, WM8990_ANTIPOP2, WM8990_BUFIOEN);
+ } else {
+ /* ON -> standby */
+
+ }
+ break;
+
+ case SND_SOC_BIAS_OFF:
+ /* Enable POBCTRL and SOFT_ST */
+ wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST |
+ WM8990_POBCTRL | WM8990_BUFIOEN);
+
+ /* Enable POBCTRL, SOFT_ST and BUFDCOPEN */
+ wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST |
+ WM8990_BUFDCOPEN | WM8990_POBCTRL |
+ WM8990_BUFIOEN);
+
+ /* mute DAC */
+ val = wm8990_read_reg_cache(codec, WM8990_DAC_CTRL);
+ wm8990_write(codec, WM8990_DAC_CTRL, val | WM8990_DAC_MUTE);
+
+ /* Enable any disabled outputs */
+ wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f03);
+
+ /* Disable VMID */
+ wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f01);
+
+ msleep(msecs_to_jiffies(300));
+
+ /* Enable all output discharge bits */
+ wm8990_write(codec, WM8990_ANTIPOP1, WM8990_DIS_LLINE |
+ WM8990_DIS_RLINE | WM8990_DIS_OUT3 |
+ WM8990_DIS_OUT4 | WM8990_DIS_LOUT |
+ WM8990_DIS_ROUT);
+
+ /* Disable VREF */
+ wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x0);
+
+ /* disable POBCTRL, SOFT_ST and BUFDCOPEN */
+ wm8990_write(codec, WM8990_ANTIPOP2, 0x0);
+ break;
+ }
+
+ codec->bias_level = level;
+ return 0;
+}
+
+#define WM8990_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000)
+
+#define WM8990_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+/*
+ * The WM8990 supports 2 different and mutually exclusive DAI
+ * configurations.
+ *
+ * 1. ADC/DAC on Primary Interface
+ * 2. ADC on Primary Interface/DAC on secondary
+ */
+struct snd_soc_dai wm8990_dai = {
+/* ADC/DAC on primary */
+ .name = "WM8990 ADC/DAC Primary",
+ .id = 1,
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8990_RATES,
+ .formats = WM8990_FORMATS,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8990_RATES,
+ .formats = WM8990_FORMATS,},
+ .ops = {
+ .hw_params = wm8990_hw_params,},
+ .dai_ops = {
+ .digital_mute = wm8990_mute,
+ .set_fmt = wm8990_set_dai_fmt,
+ .set_clkdiv = wm8990_set_dai_clkdiv,
+ .set_pll = wm8990_set_dai_pll,
+ .set_sysclk = wm8990_set_dai_sysclk,
+ },
+};
+EXPORT_SYMBOL_GPL(wm8990_dai);
+
+static int wm8990_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;
+
+ /* we only need to suspend if we are a valid card */
+ if (!codec->card)
+ return 0;
+
+ wm8990_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int wm8990_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ int i;
+ u8 data[2];
+ u16 *cache = codec->reg_cache;
+
+ /* we only need to resume if we are a valid card */
+ if (!codec->card)
+ return 0;
+
+ /* Sync reg_cache with the hardware */
+ for (i = 0; i < ARRAY_SIZE(wm8990_reg); i++) {
+ if (i + 1 == WM8990_RESET)
+ continue;
+ data[0] = ((i + 1) << 1) | ((cache[i] >> 8) & 0x0001);
+ data[1] = cache[i] & 0x00ff;
+ codec->hw_write(codec->control_data, data, 2);
+ }
+
+ wm8990_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ return 0;
+}
+
+/*
+ * initialise the WM8990 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int wm8990_init(struct snd_soc_device *socdev)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ u16 reg;
+ int ret = 0;
+
+ codec->name = "WM8990";
+ codec->owner = THIS_MODULE;
+ codec->read = wm8990_read_reg_cache;
+ codec->write = wm8990_write;
+ codec->set_bias_level = wm8990_set_bias_level;
+ codec->dai = &wm8990_dai;
+ codec->num_dai = 2;
+ codec->reg_cache_size = ARRAY_SIZE(wm8990_reg);
+ codec->reg_cache = kmemdup(wm8990_reg, sizeof(wm8990_reg), GFP_KERNEL);
+
+ if (codec->reg_cache == NULL)
+ return -ENOMEM;
+
+ wm8990_reset(codec);
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ printk(KERN_ERR "wm8990: failed to create pcms\n");
+ goto pcm_err;
+ }
+
+ /* charge output caps */
+ codec->bias_level = SND_SOC_BIAS_OFF;
+ wm8990_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ reg = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_4);
+ wm8990_write(codec, WM8990_AUDIO_INTERFACE_4, reg | WM8990_ALRCGPIO1);
+
+ reg = wm8990_read_reg_cache(codec, WM8990_GPIO1_GPIO2) &
+ ~WM8990_GPIO1_SEL_MASK;
+ wm8990_write(codec, WM8990_GPIO1_GPIO2, reg | 1);
+
+ reg = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_2);
+ wm8990_write(codec, WM8990_POWER_MANAGEMENT_2, reg | WM8990_OPCLK_ENA);
+
+ wm8990_write(codec, WM8990_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8));
+ wm8990_write(codec, WM8990_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8));
+
+ wm8990_add_controls(codec);
+ wm8990_add_widgets(codec);
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "wm8990: 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;
+}
+
+/* If the i2c layer weren't so broken, we could pass this kind of data
+ around */
+static struct snd_soc_device *wm8990_socdev;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+
+/*
+ * WM891 2 wire address is determined by GPIO5
+ * state during powerup.
+ * low = 0x34
+ * high = 0x36
+ */
+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
+
+/* Magic definition of all other variables and things */
+I2C_CLIENT_INSMOD;
+
+static struct i2c_driver wm8990_i2c_driver;
+static struct i2c_client client_template;
+
+static int wm8990_codec_probe(struct i2c_adapter *adap, int addr, int kind)
+{
+ struct snd_soc_device *socdev = wm8990_socdev;
+ struct wm8990_setup_data *setup = socdev->codec_data;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct i2c_client *i2c;
+ int ret;
+
+ if (addr != setup->i2c_address)
+ return -ENODEV;
+
+ client_template.adapter = adap;
+ client_template.addr = addr;
+
+ i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL);
+ if (i2c == NULL) {
+ kfree(codec);
+ return -ENOMEM;
+ }
+ i2c_set_clientdata(i2c, codec);
+ codec->control_data = i2c;
+
+ ret = i2c_attach_client(i2c);
+ if (ret < 0) {
+ pr_err("failed to attach codec at addr %x\n", addr);
+ goto err;
+ }
+
+ ret = wm8990_init(socdev);
+ if (ret < 0) {
+ pr_err("failed to initialise WM8990\n");
+ goto err;
+ }
+ return ret;
+
+err:
+ kfree(codec);
+ kfree(i2c);
+ return ret;
+}
+
+static int wm8990_i2c_detach(struct i2c_client *client)
+{
+ struct snd_soc_codec *codec = i2c_get_clientdata(client);
+ i2c_detach_client(client);
+ kfree(codec->reg_cache);
+ kfree(client);
+ return 0;
+}
+
+static int wm8990_i2c_attach(struct i2c_adapter *adap)
+{
+ return i2c_probe(adap, &addr_data, wm8990_codec_probe);
+}
+
+static struct i2c_driver wm8990_i2c_driver = {
+ .driver = {
+ .name = "WM8990 I2C Codec",
+ .owner = THIS_MODULE,
+ },
+ .attach_adapter = wm8990_i2c_attach,
+ .detach_client = wm8990_i2c_detach,
+ .command = NULL,
+};
+
+static struct i2c_client client_template = {
+ .name = "WM8990",
+ .driver = &wm8990_i2c_driver,
+};
+#endif
+
+static int wm8990_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct wm8990_setup_data *setup;
+ struct snd_soc_codec *codec;
+ struct wm8990_priv *wm8990;
+ int ret = 0;
+
+ 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)
+ return -ENOMEM;
+
+ wm8990 = kzalloc(sizeof(struct wm8990_priv), GFP_KERNEL);
+ if (wm8990 == NULL) {
+ kfree(codec);
+ return -ENOMEM;
+ }
+
+ codec->private_data = wm8990;
+ socdev->codec = codec;
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+ wm8990_socdev = socdev;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ if (setup->i2c_address) {
+ normal_i2c[0] = setup->i2c_address;
+ codec->hw_write = (hw_write_t)i2c_master_send;
+ ret = i2c_add_driver(&wm8990_i2c_driver);
+ if (ret != 0)
+ printk(KERN_ERR "can't add i2c driver");
+ }
+#else
+ /* Add other interfaces here */
+#endif
+ return ret;
+}
+
+/* power down chip */
+static int wm8990_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)
+ wm8990_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(&wm8990_i2c_driver);
+#endif
+ kfree(codec->private_data);
+ kfree(codec);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8990 = {
+ .probe = wm8990_probe,
+ .remove = wm8990_remove,
+ .suspend = wm8990_suspend,
+ .resume = wm8990_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8990);
+
+MODULE_DESCRIPTION("ASoC WM8990 driver");
+MODULE_AUTHOR("Liam Girdwood");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8990.h b/sound/soc/codecs/wm8990.h
new file mode 100644
index 00000000000..6bea5748528
--- /dev/null
+++ b/sound/soc/codecs/wm8990.h
@@ -0,0 +1,832 @@
+/*
+ * wm8990.h -- audio driver for WM8990
+ *
+ * Copyright 2007 Wolfson Microelectronics PLC.
+ * Author: Graeme Gregory
+ * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#ifndef __WM8990REGISTERDEFS_H__
+#define __WM8990REGISTERDEFS_H__
+
+/*
+ * Register values.
+ */
+#define WM8990_RESET 0x00
+#define WM8990_POWER_MANAGEMENT_1 0x01
+#define WM8990_POWER_MANAGEMENT_2 0x02
+#define WM8990_POWER_MANAGEMENT_3 0x03
+#define WM8990_AUDIO_INTERFACE_1 0x04
+#define WM8990_AUDIO_INTERFACE_2 0x05
+#define WM8990_CLOCKING_1 0x06
+#define WM8990_CLOCKING_2 0x07
+#define WM8990_AUDIO_INTERFACE_3 0x08
+#define WM8990_AUDIO_INTERFACE_4 0x09
+#define WM8990_DAC_CTRL 0x0A
+#define WM8990_LEFT_DAC_DIGITAL_VOLUME 0x0B
+#define WM8990_RIGHT_DAC_DIGITAL_VOLUME 0x0C
+#define WM8990_DIGITAL_SIDE_TONE 0x0D
+#define WM8990_ADC_CTRL 0x0E
+#define WM8990_LEFT_ADC_DIGITAL_VOLUME 0x0F
+#define WM8990_RIGHT_ADC_DIGITAL_VOLUME 0x10
+#define WM8990_GPIO_CTRL_1 0x12
+#define WM8990_GPIO1_GPIO2 0x13
+#define WM8990_GPIO3_GPIO4 0x14
+#define WM8990_GPIO5_GPIO6 0x15
+#define WM8990_GPIOCTRL_2 0x16
+#define WM8990_GPIO_POL 0x17
+#define WM8990_LEFT_LINE_INPUT_1_2_VOLUME 0x18
+#define WM8990_LEFT_LINE_INPUT_3_4_VOLUME 0x19
+#define WM8990_RIGHT_LINE_INPUT_1_2_VOLUME 0x1A
+#define WM8990_RIGHT_LINE_INPUT_3_4_VOLUME 0x1B
+#define WM8990_LEFT_OUTPUT_VOLUME 0x1C
+#define WM8990_RIGHT_OUTPUT_VOLUME 0x1D
+#define WM8990_LINE_OUTPUTS_VOLUME 0x1E
+#define WM8990_OUT3_4_VOLUME 0x1F
+#define WM8990_LEFT_OPGA_VOLUME 0x20
+#define WM8990_RIGHT_OPGA_VOLUME 0x21
+#define WM8990_SPEAKER_VOLUME 0x22
+#define WM8990_CLASSD1 0x23
+#define WM8990_CLASSD3 0x25
+#define WM8990_INPUT_MIXER1 0x27
+#define WM8990_INPUT_MIXER2 0x28
+#define WM8990_INPUT_MIXER3 0x29
+#define WM8990_INPUT_MIXER4 0x2A
+#define WM8990_INPUT_MIXER5 0x2B
+#define WM8990_INPUT_MIXER6 0x2C
+#define WM8990_OUTPUT_MIXER1 0x2D
+#define WM8990_OUTPUT_MIXER2 0x2E
+#define WM8990_OUTPUT_MIXER3 0x2F
+#define WM8990_OUTPUT_MIXER4 0x30
+#define WM8990_OUTPUT_MIXER5 0x31
+#define WM8990_OUTPUT_MIXER6 0x32
+#define WM8990_OUT3_4_MIXER 0x33
+#define WM8990_LINE_MIXER1 0x34
+#define WM8990_LINE_MIXER2 0x35
+#define WM8990_SPEAKER_MIXER 0x36
+#define WM8990_ADDITIONAL_CONTROL 0x37
+#define WM8990_ANTIPOP1 0x38
+#define WM8990_ANTIPOP2 0x39
+#define WM8990_MICBIAS 0x3A
+#define WM8990_PLL1 0x3C
+#define WM8990_PLL2 0x3D
+#define WM8990_PLL3 0x3E
+#define WM8990_INTDRIVBITS 0x3F
+
+#define WM8990_REGISTER_COUNT 60
+#define WM8990_MAX_REGISTER 0x3F
+
+/*
+ * Field Definitions.
+ */
+
+/*
+ * R0 (0x00) - Reset
+ */
+#define WM8990_SW_RESET_CHIP_ID_MASK 0xFFFF /* SW_RESET_CHIP_ID */
+
+/*
+ * R1 (0x01) - Power Management (1)
+ */
+#define WM8990_SPK_ENA 0x1000 /* SPK_ENA */
+#define WM8990_SPK_ENA_BIT 12
+#define WM8990_OUT3_ENA 0x0800 /* OUT3_ENA */
+#define WM8990_OUT3_ENA_BIT 11
+#define WM8990_OUT4_ENA 0x0400 /* OUT4_ENA */
+#define WM8990_OUT4_ENA_BIT 10
+#define WM8990_LOUT_ENA 0x0200 /* LOUT_ENA */
+#define WM8990_LOUT_ENA_BIT 9
+#define WM8990_ROUT_ENA 0x0100 /* ROUT_ENA */
+#define WM8990_ROUT_ENA_BIT 8
+#define WM8990_MICBIAS_ENA 0x0010 /* MICBIAS_ENA */
+#define WM8990_MICBIAS_ENA_BIT 4
+#define WM8990_VMID_MODE_MASK 0x0006 /* VMID_MODE - [2:1] */
+#define WM8990_VREF_ENA 0x0001 /* VREF_ENA */
+#define WM8990_VREF_ENA_BIT 0
+
+/*
+ * R2 (0x02) - Power Management (2)
+ */
+#define WM8990_PLL_ENA 0x8000 /* PLL_ENA */
+#define WM8990_PLL_ENA_BIT 15
+#define WM8990_TSHUT_ENA 0x4000 /* TSHUT_ENA */
+#define WM8990_TSHUT_ENA_BIT 14
+#define WM8990_TSHUT_OPDIS 0x2000 /* TSHUT_OPDIS */
+#define WM8990_TSHUT_OPDIS_BIT 13
+#define WM8990_OPCLK_ENA 0x0800 /* OPCLK_ENA */
+#define WM8990_OPCLK_ENA_BIT 11
+#define WM8990_AINL_ENA 0x0200 /* AINL_ENA */
+#define WM8990_AINL_ENA_BIT 9
+#define WM8990_AINR_ENA 0x0100 /* AINR_ENA */
+#define WM8990_AINR_ENA_BIT 8
+#define WM8990_LIN34_ENA 0x0080 /* LIN34_ENA */
+#define WM8990_LIN34_ENA_BIT 7
+#define WM8990_LIN12_ENA 0x0040 /* LIN12_ENA */
+#define WM8990_LIN12_ENA_BIT 6
+#define WM8990_RIN34_ENA 0x0020 /* RIN34_ENA */
+#define WM8990_RIN34_ENA_BIT 5
+#define WM8990_RIN12_ENA 0x0010 /* RIN12_ENA */
+#define WM8990_RIN12_ENA_BIT 4
+#define WM8990_ADCL_ENA 0x0002 /* ADCL_ENA */
+#define WM8990_ADCL_ENA_BIT 1
+#define WM8990_ADCR_ENA 0x0001 /* ADCR_ENA */
+#define WM8990_ADCR_ENA_BIT 0
+
+/*
+ * R3 (0x03) - Power Management (3)
+ */
+#define WM8990_LON_ENA 0x2000 /* LON_ENA */
+#define WM8990_LON_ENA_BIT 13
+#define WM8990_LOP_ENA 0x1000 /* LOP_ENA */
+#define WM8990_LOP_ENA_BIT 12
+#define WM8990_RON_ENA 0x0800 /* RON_ENA */
+#define WM8990_RON_ENA_BIT 11
+#define WM8990_ROP_ENA 0x0400 /* ROP_ENA */
+#define WM8990_ROP_ENA_BIT 10
+#define WM8990_LOPGA_ENA 0x0080 /* LOPGA_ENA */
+#define WM8990_LOPGA_ENA_BIT 7
+#define WM8990_ROPGA_ENA 0x0040 /* ROPGA_ENA */
+#define WM8990_ROPGA_ENA_BIT 6
+#define WM8990_LOMIX_ENA 0x0020 /* LOMIX_ENA */
+#define WM8990_LOMIX_ENA_BIT 5
+#define WM8990_ROMIX_ENA 0x0010 /* ROMIX_ENA */
+#define WM8990_ROMIX_ENA_BIT 4
+#define WM8990_DACL_ENA 0x0002 /* DACL_ENA */
+#define WM8990_DACL_ENA_BIT 1
+#define WM8990_DACR_ENA 0x0001 /* DACR_ENA */
+#define WM8990_DACR_ENA_BIT 0
+
+/*
+ * R4 (0x04) - Audio Interface (1)
+ */
+#define WM8990_AIFADCL_SRC 0x8000 /* AIFADCL_SRC */
+#define WM8990_AIFADCR_SRC 0x4000 /* AIFADCR_SRC */
+#define WM8990_AIFADC_TDM 0x2000 /* AIFADC_TDM */
+#define WM8990_AIFADC_TDM_CHAN 0x1000 /* AIFADC_TDM_CHAN */
+#define WM8990_AIF_BCLK_INV 0x0100 /* AIF_BCLK_INV */
+#define WM8990_AIF_LRCLK_INV 0x0080 /* AIF_LRCLK_INV */
+#define WM8990_AIF_WL_MASK 0x0060 /* AIF_WL - [6:5] */
+#define WM8990_AIF_WL_16BITS (0 << 5)
+#define WM8990_AIF_WL_20BITS (1 << 5)
+#define WM8990_AIF_WL_24BITS (2 << 5)
+#define WM8990_AIF_WL_32BITS (3 << 5)
+#define WM8990_AIF_FMT_MASK 0x0018 /* AIF_FMT - [4:3] */
+#define WM8990_AIF_TMF_RIGHTJ (0 << 3)
+#define WM8990_AIF_TMF_LEFTJ (1 << 3)
+#define WM8990_AIF_TMF_I2S (2 << 3)
+#define WM8990_AIF_TMF_DSP (3 << 3)
+
+/*
+ * R5 (0x05) - Audio Interface (2)
+ */
+#define WM8990_DACL_SRC 0x8000 /* DACL_SRC */
+#define WM8990_DACR_SRC 0x4000 /* DACR_SRC */
+#define WM8990_AIFDAC_TDM 0x2000 /* AIFDAC_TDM */
+#define WM8990_AIFDAC_TDM_CHAN 0x1000 /* AIFDAC_TDM_CHAN */
+#define WM8990_DAC_BOOST_MASK 0x0C00 /* DAC_BOOST */
+#define WM8990_DAC_COMP 0x0010 /* DAC_COMP */
+#define WM8990_DAC_COMPMODE 0x0008 /* DAC_COMPMODE */
+#define WM8990_ADC_COMP 0x0004 /* ADC_COMP */
+#define WM8990_ADC_COMPMODE 0x0002 /* ADC_COMPMODE */
+#define WM8990_LOOPBACK 0x0001 /* LOOPBACK */
+
+/*
+ * R6 (0x06) - Clocking (1)
+ */
+#define WM8990_TOCLK_RATE 0x8000 /* TOCLK_RATE */
+#define WM8990_TOCLK_ENA 0x4000 /* TOCLK_ENA */
+#define WM8990_OPCLKDIV_MASK 0x1E00 /* OPCLKDIV - [12:9] */
+#define WM8990_DCLKDIV_MASK 0x01C0 /* DCLKDIV - [8:6] */
+#define WM8990_BCLK_DIV_MASK 0x001E /* BCLK_DIV - [4:1] */
+#define WM8990_BCLK_DIV_1 (0x0 << 1)
+#define WM8990_BCLK_DIV_1_5 (0x1 << 1)
+#define WM8990_BCLK_DIV_2 (0x2 << 1)
+#define WM8990_BCLK_DIV_3 (0x3 << 1)
+#define WM8990_BCLK_DIV_4 (0x4 << 1)
+#define WM8990_BCLK_DIV_5_5 (0x5 << 1)
+#define WM8990_BCLK_DIV_6 (0x6 << 1)
+#define WM8990_BCLK_DIV_8 (0x7 << 1)
+#define WM8990_BCLK_DIV_11 (0x8 << 1)
+#define WM8990_BCLK_DIV_12 (0x9 << 1)
+#define WM8990_BCLK_DIV_16 (0xA << 1)
+#define WM8990_BCLK_DIV_22 (0xB << 1)
+#define WM8990_BCLK_DIV_24 (0xC << 1)
+#define WM8990_BCLK_DIV_32 (0xD << 1)
+#define WM8990_BCLK_DIV_44 (0xE << 1)
+#define WM8990_BCLK_DIV_48 (0xF << 1)
+
+/*
+ * R7 (0x07) - Clocking (2)
+ */
+#define WM8990_MCLK_SRC 0x8000 /* MCLK_SRC */
+#define WM8990_SYSCLK_SRC 0x4000 /* SYSCLK_SRC */
+#define WM8990_CLK_FORCE 0x2000 /* CLK_FORCE */
+#define WM8990_MCLK_DIV_MASK 0x1800 /* MCLK_DIV - [12:11] */
+#define WM8990_MCLK_DIV_1 (0 << 11)
+#define WM8990_MCLK_DIV_2 (2 << 11)
+#define WM8990_MCLK_INV 0x0400 /* MCLK_INV */
+#define WM8990_ADC_CLKDIV_MASK 0x00E0 /* ADC_CLKDIV */
+#define WM8990_ADC_CLKDIV_1 (0 << 5)
+#define WM8990_ADC_CLKDIV_1_5 (1 << 5)
+#define WM8990_ADC_CLKDIV_2 (2 << 5)
+#define WM8990_ADC_CLKDIV_3 (3 << 5)
+#define WM8990_ADC_CLKDIV_4 (4 << 5)
+#define WM8990_ADC_CLKDIV_5_5 (5 << 5)
+#define WM8990_ADC_CLKDIV_6 (6 << 5)
+#define WM8990_DAC_CLKDIV_MASK 0x001C /* DAC_CLKDIV - [4:2] */
+#define WM8990_DAC_CLKDIV_1 (0 << 2)
+#define WM8990_DAC_CLKDIV_1_5 (1 << 2)
+#define WM8990_DAC_CLKDIV_2 (2 << 2)
+#define WM8990_DAC_CLKDIV_3 (3 << 2)
+#define WM8990_DAC_CLKDIV_4 (4 << 2)
+#define WM8990_DAC_CLKDIV_5_5 (5 << 2)
+#define WM8990_DAC_CLKDIV_6 (6 << 2)
+
+/*
+ * R8 (0x08) - Audio Interface (3)
+ */
+#define WM8990_AIF_MSTR1 0x8000 /* AIF_MSTR1 */
+#define WM8990_AIF_MSTR2 0x4000 /* AIF_MSTR2 */
+#define WM8990_AIF_SEL 0x2000 /* AIF_SEL */
+#define WM8990_ADCLRC_DIR 0x0800 /* ADCLRC_DIR */
+#define WM8990_ADCLRC_RATE_MASK 0x07FF /* ADCLRC_RATE */
+
+/*
+ * R9 (0x09) - Audio Interface (4)
+ */
+#define WM8990_ALRCGPIO1 0x8000 /* ALRCGPIO1 */
+#define WM8990_ALRCBGPIO6 0x4000 /* ALRCBGPIO6 */
+#define WM8990_AIF_TRIS 0x2000 /* AIF_TRIS */
+#define WM8990_DACLRC_DIR 0x0800 /* DACLRC_DIR */
+#define WM8990_DACLRC_RATE_MASK 0x07FF /* DACLRC_RATE */
+
+/*
+ * R10 (0x0A) - DAC CTRL
+ */
+#define WM8990_AIF_LRCLKRATE 0x0400 /* AIF_LRCLKRATE */
+#define WM8990_DAC_MONO 0x0200 /* DAC_MONO */
+#define WM8990_DAC_SB_FILT 0x0100 /* DAC_SB_FILT */
+#define WM8990_DAC_MUTERATE 0x0080 /* DAC_MUTERATE */
+#define WM8990_DAC_MUTEMODE 0x0040 /* DAC_MUTEMODE */
+#define WM8990_DEEMP_MASK 0x0030 /* DEEMP - [5:4] */
+#define WM8990_DAC_MUTE 0x0004 /* DAC_MUTE */
+#define WM8990_DACL_DATINV 0x0002 /* DACL_DATINV */
+#define WM8990_DACR_DATINV 0x0001 /* DACR_DATINV */
+
+/*
+ * R11 (0x0B) - Left DAC Digital Volume
+ */
+#define WM8990_DAC_VU 0x0100 /* DAC_VU */
+#define WM8990_DACL_VOL_MASK 0x00FF /* DACL_VOL - [7:0] */
+#define WM8990_DACL_VOL_SHIFT 0
+/*
+ * R12 (0x0C) - Right DAC Digital Volume
+ */
+#define WM8990_DAC_VU 0x0100 /* DAC_VU */
+#define WM8990_DACR_VOL_MASK 0x00FF /* DACR_VOL - [7:0] */
+#define WM8990_DACR_VOL_SHIFT 0
+/*
+ * R13 (0x0D) - Digital Side Tone
+ */
+#define WM8990_ADCL_DAC_SVOL_MASK 0x0F /* ADCL_DAC_SVOL */
+#define WM8990_ADCL_DAC_SVOL_SHIFT 9
+#define WM8990_ADCR_DAC_SVOL_MASK 0x0F /* ADCR_DAC_SVOL */
+#define WM8990_ADCR_DAC_SVOL_SHIFT 5
+#define WM8990_ADC_TO_DACL_MASK 0x03 /* ADC_TO_DACL - [3:2] */
+#define WM8990_ADC_TO_DACL_SHIFT 2
+#define WM8990_ADC_TO_DACR_MASK 0x03 /* ADC_TO_DACR - [1:0] */
+#define WM8990_ADC_TO_DACR_SHIFT 0
+
+/*
+ * R14 (0x0E) - ADC CTRL
+ */
+#define WM8990_ADC_HPF_ENA 0x0100 /* ADC_HPF_ENA */
+#define WM8990_ADC_HPF_ENA_BIT 8
+#define WM8990_ADC_HPF_CUT_MASK 0x03 /* ADC_HPF_CUT - [6:5] */
+#define WM8990_ADC_HPF_CUT_SHIFT 5
+#define WM8990_ADCL_DATINV 0x0002 /* ADCL_DATINV */
+#define WM8990_ADCL_DATINV_BIT 1
+#define WM8990_ADCR_DATINV 0x0001 /* ADCR_DATINV */
+#define WM8990_ADCR_DATINV_BIT 0
+
+/*
+ * R15 (0x0F) - Left ADC Digital Volume
+ */
+#define WM8990_ADC_VU 0x0100 /* ADC_VU */
+#define WM8990_ADCL_VOL_MASK 0x00FF /* ADCL_VOL - [7:0] */
+#define WM8990_ADCL_VOL_SHIFT 0
+
+/*
+ * R16 (0x10) - Right ADC Digital Volume
+ */
+#define WM8990_ADC_VU 0x0100 /* ADC_VU */
+#define WM8990_ADCR_VOL_MASK 0x00FF /* ADCR_VOL - [7:0] */
+#define WM8990_ADCR_VOL_SHIFT 0
+
+/*
+ * R18 (0x12) - GPIO CTRL 1
+ */
+#define WM8990_IRQ 0x1000 /* IRQ */
+#define WM8990_TEMPOK 0x0800 /* TEMPOK */
+#define WM8990_MICSHRT 0x0400 /* MICSHRT */
+#define WM8990_MICDET 0x0200 /* MICDET */
+#define WM8990_PLL_LCK 0x0100 /* PLL_LCK */
+#define WM8990_GPI8_STATUS 0x0080 /* GPI8_STATUS */
+#define WM8990_GPI7_STATUS 0x0040 /* GPI7_STATUS */
+#define WM8990_GPIO6_STATUS 0x0020 /* GPIO6_STATUS */
+#define WM8990_GPIO5_STATUS 0x0010 /* GPIO5_STATUS */
+#define WM8990_GPIO4_STATUS 0x0008 /* GPIO4_STATUS */
+#define WM8990_GPIO3_STATUS 0x0004 /* GPIO3_STATUS */
+#define WM8990_GPIO2_STATUS 0x0002 /* GPIO2_STATUS */
+#define WM8990_GPIO1_STATUS 0x0001 /* GPIO1_STATUS */
+
+/*
+ * R19 (0x13) - GPIO1 & GPIO2
+ */
+#define WM8990_GPIO2_DEB_ENA 0x8000 /* GPIO2_DEB_ENA */
+#define WM8990_GPIO2_IRQ_ENA 0x4000 /* GPIO2_IRQ_ENA */
+#define WM8990_GPIO2_PU 0x2000 /* GPIO2_PU */
+#define WM8990_GPIO2_PD 0x1000 /* GPIO2_PD */
+#define WM8990_GPIO2_SEL_MASK 0x0F00 /* GPIO2_SEL - [11:8] */
+#define WM8990_GPIO1_DEB_ENA 0x0080 /* GPIO1_DEB_ENA */
+#define WM8990_GPIO1_IRQ_ENA 0x0040 /* GPIO1_IRQ_ENA */
+#define WM8990_GPIO1_PU 0x0020 /* GPIO1_PU */
+#define WM8990_GPIO1_PD 0x0010 /* GPIO1_PD */
+#define WM8990_GPIO1_SEL_MASK 0x000F /* GPIO1_SEL - [3:0] */
+
+/*
+ * R20 (0x14) - GPIO3 & GPIO4
+ */
+#define WM8990_GPIO4_DEB_ENA 0x8000 /* GPIO4_DEB_ENA */
+#define WM8990_GPIO4_IRQ_ENA 0x4000 /* GPIO4_IRQ_ENA */
+#define WM8990_GPIO4_PU 0x2000 /* GPIO4_PU */
+#define WM8990_GPIO4_PD 0x1000 /* GPIO4_PD */
+#define WM8990_GPIO4_SEL_MASK 0x0F00 /* GPIO4_SEL - [11:8] */
+#define WM8990_GPIO3_DEB_ENA 0x0080 /* GPIO3_DEB_ENA */
+#define WM8990_GPIO3_IRQ_ENA 0x0040 /* GPIO3_IRQ_ENA */
+#define WM8990_GPIO3_PU 0x0020 /* GPIO3_PU */
+#define WM8990_GPIO3_PD 0x0010 /* GPIO3_PD */
+#define WM8990_GPIO3_SEL_MASK 0x000F /* GPIO3_SEL - [3:0] */
+
+/*
+ * R21 (0x15) - GPIO5 & GPIO6
+ */
+#define WM8990_GPIO6_DEB_ENA 0x8000 /* GPIO6_DEB_ENA */
+#define WM8990_GPIO6_IRQ_ENA 0x4000 /* GPIO6_IRQ_ENA */
+#define WM8990_GPIO6_PU 0x2000 /* GPIO6_PU */
+#define WM8990_GPIO6_PD 0x1000 /* GPIO6_PD */
+#define WM8990_GPIO6_SEL_MASK 0x0F00 /* GPIO6_SEL - [11:8] */
+#define WM8990_GPIO5_DEB_ENA 0x0080 /* GPIO5_DEB_ENA */
+#define WM8990_GPIO5_IRQ_ENA 0x0040 /* GPIO5_IRQ_ENA */
+#define WM8990_GPIO5_PU 0x0020 /* GPIO5_PU */
+#define WM8990_GPIO5_PD 0x0010 /* GPIO5_PD */
+#define WM8990_GPIO5_SEL_MASK 0x000F /* GPIO5_SEL - [3:0] */
+
+/*
+ * R22 (0x16) - GPIOCTRL 2
+ */
+#define WM8990_RD_3W_ENA 0x8000 /* RD_3W_ENA */
+#define WM8990_MODE_3W4W 0x4000 /* MODE_3W4W */
+#define WM8990_TEMPOK_IRQ_ENA 0x0800 /* TEMPOK_IRQ_ENA */
+#define WM8990_MICSHRT_IRQ_ENA 0x0400 /* MICSHRT_IRQ_ENA */
+#define WM8990_MICDET_IRQ_ENA 0x0200 /* MICDET_IRQ_ENA */
+#define WM8990_PLL_LCK_IRQ_ENA 0x0100 /* PLL_LCK_IRQ_ENA */
+#define WM8990_GPI8_DEB_ENA 0x0080 /* GPI8_DEB_ENA */
+#define WM8990_GPI8_IRQ_ENA 0x0040 /* GPI8_IRQ_ENA */
+#define WM8990_GPI8_ENA 0x0010 /* GPI8_ENA */
+#define WM8990_GPI7_DEB_ENA 0x0008 /* GPI7_DEB_ENA */
+#define WM8990_GPI7_IRQ_ENA 0x0004 /* GPI7_IRQ_ENA */
+#define WM8990_GPI7_ENA 0x0001 /* GPI7_ENA */
+
+/*
+ * R23 (0x17) - GPIO_POL
+ */
+#define WM8990_IRQ_INV 0x1000 /* IRQ_INV */
+#define WM8990_TEMPOK_POL 0x0800 /* TEMPOK_POL */
+#define WM8990_MICSHRT_POL 0x0400 /* MICSHRT_POL */
+#define WM8990_MICDET_POL 0x0200 /* MICDET_POL */
+#define WM8990_PLL_LCK_POL 0x0100 /* PLL_LCK_POL */
+#define WM8990_GPI8_POL 0x0080 /* GPI8_POL */
+#define WM8990_GPI7_POL 0x0040 /* GPI7_POL */
+#define WM8990_GPIO6_POL 0x0020 /* GPIO6_POL */
+#define WM8990_GPIO5_POL 0x0010 /* GPIO5_POL */
+#define WM8990_GPIO4_POL 0x0008 /* GPIO4_POL */
+#define WM8990_GPIO3_POL 0x0004 /* GPIO3_POL */
+#define WM8990_GPIO2_POL 0x0002 /* GPIO2_POL */
+#define WM8990_GPIO1_POL 0x0001 /* GPIO1_POL */
+
+/*
+ * R24 (0x18) - Left Line Input 1&2 Volume
+ */
+#define WM8990_IPVU 0x0100 /* IPVU */
+#define WM8990_LI12MUTE 0x0080 /* LI12MUTE */
+#define WM8990_LI12MUTE_BIT 7
+#define WM8990_LI12ZC 0x0040 /* LI12ZC */
+#define WM8990_LI12ZC_BIT 6
+#define WM8990_LIN12VOL_MASK 0x001F /* LIN12VOL - [4:0] */
+#define WM8990_LIN12VOL_SHIFT 0
+/*
+ * R25 (0x19) - Left Line Input 3&4 Volume
+ */
+#define WM8990_IPVU 0x0100 /* IPVU */
+#define WM8990_LI34MUTE 0x0080 /* LI34MUTE */
+#define WM8990_LI34MUTE_BIT 7
+#define WM8990_LI34ZC 0x0040 /* LI34ZC */
+#define WM8990_LI34ZC_BIT 6
+#define WM8990_LIN34VOL_MASK 0x001F /* LIN34VOL - [4:0] */
+#define WM8990_LIN34VOL_SHIFT 0
+
+/*
+ * R26 (0x1A) - Right Line Input 1&2 Volume
+ */
+#define WM8990_IPVU 0x0100 /* IPVU */
+#define WM8990_RI12MUTE 0x0080 /* RI12MUTE */
+#define WM8990_RI12MUTE_BIT 7
+#define WM8990_RI12ZC 0x0040 /* RI12ZC */
+#define WM8990_RI12ZC_BIT 6
+#define WM8990_RIN12VOL_MASK 0x001F /* RIN12VOL - [4:0] */
+#define WM8990_RIN12VOL_SHIFT 0
+
+/*
+ * R27 (0x1B) - Right Line Input 3&4 Volume
+ */
+#define WM8990_IPVU 0x0100 /* IPVU */
+#define WM8990_RI34MUTE 0x0080 /* RI34MUTE */
+#define WM8990_RI34MUTE_BIT 7
+#define WM8990_RI34ZC 0x0040 /* RI34ZC */
+#define WM8990_RI34ZC_BIT 6
+#define WM8990_RIN34VOL_MASK 0x001F /* RIN34VOL - [4:0] */
+#define WM8990_RIN34VOL_SHIFT 0
+
+/*
+ * R28 (0x1C) - Left Output Volume
+ */
+#define WM8990_OPVU 0x0100 /* OPVU */
+#define WM8990_LOZC 0x0080 /* LOZC */
+#define WM8990_LOZC_BIT 7
+#define WM8990_LOUTVOL_MASK 0x007F /* LOUTVOL - [6:0] */
+#define WM8990_LOUTVOL_SHIFT 0
+/*
+ * R29 (0x1D) - Right Output Volume
+ */
+#define WM8990_OPVU 0x0100 /* OPVU */
+#define WM8990_ROZC 0x0080 /* ROZC */
+#define WM8990_ROZC_BIT 7
+#define WM8990_ROUTVOL_MASK 0x007F /* ROUTVOL - [6:0] */
+#define WM8990_ROUTVOL_SHIFT 0
+/*
+ * R30 (0x1E) - Line Outputs Volume
+ */
+#define WM8990_LONMUTE 0x0040 /* LONMUTE */
+#define WM8990_LONMUTE_BIT 6
+#define WM8990_LOPMUTE 0x0020 /* LOPMUTE */
+#define WM8990_LOPMUTE_BIT 5
+#define WM8990_LOATTN 0x0010 /* LOATTN */
+#define WM8990_LOATTN_BIT 4
+#define WM8990_RONMUTE 0x0004 /* RONMUTE */
+#define WM8990_RONMUTE_BIT 2
+#define WM8990_ROPMUTE 0x0002 /* ROPMUTE */
+#define WM8990_ROPMUTE_BIT 1
+#define WM8990_ROATTN 0x0001 /* ROATTN */
+#define WM8990_ROATTN_BIT 0
+
+/*
+ * R31 (0x1F) - Out3/4 Volume
+ */
+#define WM8990_OUT3MUTE 0x0020 /* OUT3MUTE */
+#define WM8990_OUT3MUTE_BIT 5
+#define WM8990_OUT3ATTN 0x0010 /* OUT3ATTN */
+#define WM8990_OUT3ATTN_BIT 4
+#define WM8990_OUT4MUTE 0x0002 /* OUT4MUTE */
+#define WM8990_OUT4MUTE_BIT 1
+#define WM8990_OUT4ATTN 0x0001 /* OUT4ATTN */
+#define WM8990_OUT4ATTN_BIT 0
+
+/*
+ * R32 (0x20) - Left OPGA Volume
+ */
+#define WM8990_OPVU 0x0100 /* OPVU */
+#define WM8990_LOPGAZC 0x0080 /* LOPGAZC */
+#define WM8990_LOPGAZC_BIT 7
+#define WM8990_LOPGAVOL_MASK 0x007F /* LOPGAVOL - [6:0] */
+#define WM8990_LOPGAVOL_SHIFT 0
+
+/*
+ * R33 (0x21) - Right OPGA Volume
+ */
+#define WM8990_OPVU 0x0100 /* OPVU */
+#define WM8990_ROPGAZC 0x0080 /* ROPGAZC */
+#define WM8990_ROPGAZC_BIT 7
+#define WM8990_ROPGAVOL_MASK 0x007F /* ROPGAVOL - [6:0] */
+#define WM8990_ROPGAVOL_SHIFT 0
+/*
+ * R34 (0x22) - Speaker Volume
+ */
+#define WM8990_SPKVOL_MASK 0x0003 /* SPKVOL - [1:0] */
+#define WM8990_SPKVOL_SHIFT 0
+
+/*
+ * R35 (0x23) - ClassD1
+ */
+#define WM8990_CDMODE 0x0100 /* CDMODE */
+#define WM8990_CDMODE_BIT 8
+
+/*
+ * R37 (0x25) - ClassD3
+ */
+#define WM8990_DCGAIN_MASK 0x0007 /* DCGAIN - [5:3] */
+#define WM8990_DCGAIN_SHIFT 3
+#define WM8990_ACGAIN_MASK 0x0007 /* ACGAIN - [2:0] */
+#define WM8990_ACGAIN_SHIFT 0
+/*
+ * R39 (0x27) - Input Mixer1
+ */
+#define WM8990_AINLMODE_MASK 0x000C /* AINLMODE - [3:2] */
+#define WM8990_AINLMODE_SHIFT 2
+#define WM8990_AINRMODE_MASK 0x0003 /* AINRMODE - [1:0] */
+#define WM8990_AINRMODE_SHIFT 0
+
+/*
+ * R40 (0x28) - Input Mixer2
+ */
+#define WM8990_LMP4 0x0080 /* LMP4 */
+#define WM8990_LMP4_BIT 7 /* LMP4 */
+#define WM8990_LMN3 0x0040 /* LMN3 */
+#define WM8990_LMN3_BIT 6 /* LMN3 */
+#define WM8990_LMP2 0x0020 /* LMP2 */
+#define WM8990_LMP2_BIT 5 /* LMP2 */
+#define WM8990_LMN1 0x0010 /* LMN1 */
+#define WM8990_LMN1_BIT 4 /* LMN1 */
+#define WM8990_RMP4 0x0008 /* RMP4 */
+#define WM8990_RMP4_BIT 3 /* RMP4 */
+#define WM8990_RMN3 0x0004 /* RMN3 */
+#define WM8990_RMN3_BIT 2 /* RMN3 */
+#define WM8990_RMP2 0x0002 /* RMP2 */
+#define WM8990_RMP2_BIT 1 /* RMP2 */
+#define WM8990_RMN1 0x0001 /* RMN1 */
+#define WM8990_RMN1_BIT 0 /* RMN1 */
+
+/*
+ * R41 (0x29) - Input Mixer3
+ */
+#define WM8990_L34MNB 0x0100 /* L34MNB */
+#define WM8990_L34MNB_BIT 8
+#define WM8990_L34MNBST 0x0080 /* L34MNBST */
+#define WM8990_L34MNBST_BIT 7
+#define WM8990_L12MNB 0x0020 /* L12MNB */
+#define WM8990_L12MNB_BIT 5
+#define WM8990_L12MNBST 0x0010 /* L12MNBST */
+#define WM8990_L12MNBST_BIT 4
+#define WM8990_LDBVOL_MASK 0x0007 /* LDBVOL - [2:0] */
+#define WM8990_LDBVOL_SHIFT 0
+
+/*
+ * R42 (0x2A) - Input Mixer4
+ */
+#define WM8990_R34MNB 0x0100 /* R34MNB */
+#define WM8990_R34MNB_BIT 8
+#define WM8990_R34MNBST 0x0080 /* R34MNBST */
+#define WM8990_R34MNBST_BIT 7
+#define WM8990_R12MNB 0x0020 /* R12MNB */
+#define WM8990_R12MNB_BIT 5
+#define WM8990_R12MNBST 0x0010 /* R12MNBST */
+#define WM8990_R12MNBST_BIT 4
+#define WM8990_RDBVOL_MASK 0x0007 /* RDBVOL - [2:0] */
+#define WM8990_RDBVOL_SHIFT 0
+
+/*
+ * R43 (0x2B) - Input Mixer5
+ */
+#define WM8990_LI2BVOL_MASK 0x07 /* LI2BVOL - [8:6] */
+#define WM8990_LI2BVOL_SHIFT 6
+#define WM8990_LR4BVOL_MASK 0x07 /* LR4BVOL - [5:3] */
+#define WM8990_LR4BVOL_SHIFT 3
+#define WM8990_LL4BVOL_MASK 0x07 /* LL4BVOL - [2:0] */
+#define WM8990_LL4BVOL_SHIFT 0
+
+/*
+ * R44 (0x2C) - Input Mixer6
+ */
+#define WM8990_RI2BVOL_MASK 0x07 /* RI2BVOL - [8:6] */
+#define WM8990_RI2BVOL_SHIFT 6
+#define WM8990_RL4BVOL_MASK 0x07 /* RL4BVOL - [5:3] */
+#define WM8990_RL4BVOL_SHIFT 3
+#define WM8990_RR4BVOL_MASK 0x07 /* RR4BVOL - [2:0] */
+#define WM8990_RR4BVOL_SHIFT 0
+
+/*
+ * R45 (0x2D) - Output Mixer1
+ */
+#define WM8990_LRBLO 0x0080 /* LRBLO */
+#define WM8990_LRBLO_BIT 7
+#define WM8990_LLBLO 0x0040 /* LLBLO */
+#define WM8990_LLBLO_BIT 6
+#define WM8990_LRI3LO 0x0020 /* LRI3LO */
+#define WM8990_LRI3LO_BIT 5
+#define WM8990_LLI3LO 0x0010 /* LLI3LO */
+#define WM8990_LLI3LO_BIT 4
+#define WM8990_LR12LO 0x0008 /* LR12LO */
+#define WM8990_LR12LO_BIT 3
+#define WM8990_LL12LO 0x0004 /* LL12LO */
+#define WM8990_LL12LO_BIT 2
+#define WM8990_LDLO 0x0001 /* LDLO */
+#define WM8990_LDLO_BIT 0
+
+/*
+ * R46 (0x2E) - Output Mixer2
+ */
+#define WM8990_RLBRO 0x0080 /* RLBRO */
+#define WM8990_RLBRO_BIT 7
+#define WM8990_RRBRO 0x0040 /* RRBRO */
+#define WM8990_RRBRO_BIT 6
+#define WM8990_RLI3RO 0x0020 /* RLI3RO */
+#define WM8990_RLI3RO_BIT 5
+#define WM8990_RRI3RO 0x0010 /* RRI3RO */
+#define WM8990_RRI3RO_BIT 4
+#define WM8990_RL12RO 0x0008 /* RL12RO */
+#define WM8990_RL12RO_BIT 3
+#define WM8990_RR12RO 0x0004 /* RR12RO */
+#define WM8990_RR12RO_BIT 2
+#define WM8990_RDRO 0x0001 /* RDRO */
+#define WM8990_RDRO_BIT 0
+
+/*
+ * R47 (0x2F) - Output Mixer3
+ */
+#define WM8990_LLI3LOVOL_MASK 0x07 /* LLI3LOVOL - [8:6] */
+#define WM8990_LLI3LOVOL_SHIFT 6
+#define WM8990_LR12LOVOL_MASK 0x07 /* LR12LOVOL - [5:3] */
+#define WM8990_LR12LOVOL_SHIFT 3
+#define WM8990_LL12LOVOL_MASK 0x07 /* LL12LOVOL - [2:0] */
+#define WM8990_LL12LOVOL_SHIFT 0
+
+/*
+ * R48 (0x30) - Output Mixer4
+ */
+#define WM8990_RRI3ROVOL_MASK 0x07 /* RRI3ROVOL - [8:6] */
+#define WM8990_RRI3ROVOL_SHIFT 6
+#define WM8990_RL12ROVOL_MASK 0x07 /* RL12ROVOL - [5:3] */
+#define WM8990_RL12ROVOL_SHIFT 3
+#define WM8990_RR12ROVOL_MASK 0x07 /* RR12ROVOL - [2:0] */
+#define WM8990_RR12ROVOL_SHIFT 0
+
+/*
+ * R49 (0x31) - Output Mixer5
+ */
+#define WM8990_LRI3LOVOL_MASK 0x07 /* LRI3LOVOL - [8:6] */
+#define WM8990_LRI3LOVOL_SHIFT 6
+#define WM8990_LRBLOVOL_MASK 0x07 /* LRBLOVOL - [5:3] */
+#define WM8990_LRBLOVOL_SHIFT 3
+#define WM8990_LLBLOVOL_MASK 0x07 /* LLBLOVOL - [2:0] */
+#define WM8990_LLBLOVOL_SHIFT 0
+
+/*
+ * R50 (0x32) - Output Mixer6
+ */
+#define WM8990_RLI3ROVOL_MASK 0x07 /* RLI3ROVOL - [8:6] */
+#define WM8990_RLI3ROVOL_SHIFT 6
+#define WM8990_RLBROVOL_MASK 0x07 /* RLBROVOL - [5:3] */
+#define WM8990_RLBROVOL_SHIFT 3
+#define WM8990_RRBROVOL_MASK 0x07 /* RRBROVOL - [2:0] */
+#define WM8990_RRBROVOL_SHIFT 0
+
+/*
+ * R51 (0x33) - Out3/4 Mixer
+ */
+#define WM8990_VSEL_MASK 0x0180 /* VSEL - [8:7] */
+#define WM8990_LI4O3 0x0020 /* LI4O3 */
+#define WM8990_LI4O3_BIT 5
+#define WM8990_LPGAO3 0x0010 /* LPGAO3 */
+#define WM8990_LPGAO3_BIT 4
+#define WM8990_RI4O4 0x0002 /* RI4O4 */
+#define WM8990_RI4O4_BIT 1
+#define WM8990_RPGAO4 0x0001 /* RPGAO4 */
+#define WM8990_RPGAO4_BIT 0
+/*
+ * R52 (0x34) - Line Mixer1
+ */
+#define WM8990_LLOPGALON 0x0040 /* LLOPGALON */
+#define WM8990_LLOPGALON_BIT 6
+#define WM8990_LROPGALON 0x0020 /* LROPGALON */
+#define WM8990_LROPGALON_BIT 5
+#define WM8990_LOPLON 0x0010 /* LOPLON */
+#define WM8990_LOPLON_BIT 4
+#define WM8990_LR12LOP 0x0004 /* LR12LOP */
+#define WM8990_LR12LOP_BIT 2
+#define WM8990_LL12LOP 0x0002 /* LL12LOP */
+#define WM8990_LL12LOP_BIT 1
+#define WM8990_LLOPGALOP 0x0001 /* LLOPGALOP */
+#define WM8990_LLOPGALOP_BIT 0
+/*
+ * R53 (0x35) - Line Mixer2
+ */
+#define WM8990_RROPGARON 0x0040 /* RROPGARON */
+#define WM8990_RROPGARON_BIT 6
+#define WM8990_RLOPGARON 0x0020 /* RLOPGARON */
+#define WM8990_RLOPGARON_BIT 5
+#define WM8990_ROPRON 0x0010 /* ROPRON */
+#define WM8990_ROPRON_BIT 4
+#define WM8990_RL12ROP 0x0004 /* RL12ROP */
+#define WM8990_RL12ROP_BIT 2
+#define WM8990_RR12ROP 0x0002 /* RR12ROP */
+#define WM8990_RR12ROP_BIT 1
+#define WM8990_RROPGAROP 0x0001 /* RROPGAROP */
+#define WM8990_RROPGAROP_BIT 0
+
+/*
+ * R54 (0x36) - Speaker Mixer
+ */
+#define WM8990_LB2SPK 0x0080 /* LB2SPK */
+#define WM8990_LB2SPK_BIT 7
+#define WM8990_RB2SPK 0x0040 /* RB2SPK */
+#define WM8990_RB2SPK_BIT 6
+#define WM8990_LI2SPK 0x0020 /* LI2SPK */
+#define WM8990_LI2SPK_BIT 5
+#define WM8990_RI2SPK 0x0010 /* RI2SPK */
+#define WM8990_RI2SPK_BIT 4
+#define WM8990_LOPGASPK 0x0008 /* LOPGASPK */
+#define WM8990_LOPGASPK_BIT 3
+#define WM8990_ROPGASPK 0x0004 /* ROPGASPK */
+#define WM8990_ROPGASPK_BIT 2
+#define WM8990_LDSPK 0x0002 /* LDSPK */
+#define WM8990_LDSPK_BIT 1
+#define WM8990_RDSPK 0x0001 /* RDSPK */
+#define WM8990_RDSPK_BIT 0
+
+/*
+ * R55 (0x37) - Additional Control
+ */
+#define WM8990_VROI 0x0001 /* VROI */
+
+/*
+ * R56 (0x38) - AntiPOP1
+ */
+#define WM8990_DIS_LLINE 0x0020 /* DIS_LLINE */
+#define WM8990_DIS_RLINE 0x0010 /* DIS_RLINE */
+#define WM8990_DIS_OUT3 0x0008 /* DIS_OUT3 */
+#define WM8990_DIS_OUT4 0x0004 /* DIS_OUT4 */
+#define WM8990_DIS_LOUT 0x0002 /* DIS_LOUT */
+#define WM8990_DIS_ROUT 0x0001 /* DIS_ROUT */
+
+/*
+ * R57 (0x39) - AntiPOP2
+ */
+#define WM8990_SOFTST 0x0040 /* SOFTST */
+#define WM8990_BUFIOEN 0x0008 /* BUFIOEN */
+#define WM8990_BUFDCOPEN 0x0004 /* BUFDCOPEN */
+#define WM8990_POBCTRL 0x0002 /* POBCTRL */
+#define WM8990_VMIDTOG 0x0001 /* VMIDTOG */
+
+/*
+ * R58 (0x3A) - MICBIAS
+ */
+#define WM8990_MCDSCTH_MASK 0x00C0 /* MCDSCTH - [7:6] */
+#define WM8990_MCDTHR_MASK 0x0038 /* MCDTHR - [5:3] */
+#define WM8990_MCD 0x0004 /* MCD */
+#define WM8990_MBSEL 0x0001 /* MBSEL */
+
+/*
+ * R60 (0x3C) - PLL1
+ */
+#define WM8990_SDM 0x0080 /* SDM */
+#define WM8990_PRESCALE 0x0040 /* PRESCALE */
+#define WM8990_PLLN_MASK 0x000F /* PLLN - [3:0] */
+
+/*
+ * R61 (0x3D) - PLL2
+ */
+#define WM8990_PLLK1_MASK 0x00FF /* PLLK1 - [7:0] */
+
+/*
+ * R62 (0x3E) - PLL3
+ */
+#define WM8990_PLLK2_MASK 0x00FF /* PLLK2 - [7:0] */
+
+/*
+ * R63 (0x3F) - Internal Driver Bits
+ */
+#define WM8990_INMIXL_PWR_BIT 0
+#define WM8990_AINLMUX_PWR_BIT 1
+#define WM8990_INMIXR_PWR_BIT 2
+#define WM8990_AINRMUX_PWR_BIT 3
+
+struct wm8990_setup_data {
+ unsigned short i2c_address;
+};
+
+#define WM8990_MCLK_DIV 0
+#define WM8990_DACCLK_DIV 1
+#define WM8990_ADCCLK_DIV 2
+#define WM8990_BCLK_DIV 3
+
+extern struct snd_soc_dai wm8990_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8990;
+
+#endif /* __WM8990REGISTERDEFS_H__ */
+/*------------------------------ END OF FILE ---------------------------------*/
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 76c1e2d33e7..9fc8edd8222 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -9,9 +9,6 @@
* 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
- * 4th Feb 2006 Initial version.
*/
#include <linux/init.h>
@@ -25,6 +22,7 @@
#include <sound/initval.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
+#include "wm9712.h"
#define WM9712_VERSION "0.4"
@@ -351,7 +349,7 @@ SND_SOC_DAPM_INPUT("MIC1"),
SND_SOC_DAPM_INPUT("MIC2"),
};
-static const char *audio_map[][3] = {
+static const struct snd_soc_dapm_route audio_map[] = {
/* virtual mixer - mixes left & right channels for spk and mono */
{"AC97 Mixer", NULL, "Left DAC"},
{"AC97 Mixer", NULL, "Right DAC"},
@@ -446,21 +444,14 @@ static const char *audio_map[][3] = {
{"Speaker PGA", NULL, "Speaker Mux"},
{"LOUT2", NULL, "Speaker PGA"},
{"ROUT2", NULL, "Speaker PGA"},
-
- {NULL, NULL, NULL},
};
static int wm9712_add_widgets(struct snd_soc_codec *codec)
{
- int i;
-
- for (i = 0; i < ARRAY_SIZE(wm9712_dapm_widgets); i++)
- snd_soc_dapm_new_control(codec, &wm9712_dapm_widgets[i]);
+ snd_soc_dapm_new_controls(codec, wm9712_dapm_widgets,
+ ARRAY_SIZE(wm9712_dapm_widgets));
- /* set up audio path connects */
- for (i = 0; audio_map[i][0] != NULL; i++)
- snd_soc_dapm_connect_input(codec, audio_map[i][0],
- audio_map[i][1], audio_map[i][2]);
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
snd_soc_dapm_new_widgets(codec);
return 0;
@@ -541,7 +532,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream)
SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\
SNDRV_PCM_RATE_48000)
-struct snd_soc_codec_dai wm9712_dai[] = {
+struct snd_soc_dai wm9712_dai[] = {
{
.name = "AC97 HiFi",
.type = SND_SOC_DAI_AC97_BUS,
@@ -574,23 +565,23 @@ struct snd_soc_codec_dai wm9712_dai[] = {
};
EXPORT_SYMBOL_GPL(wm9712_dai);
-static int wm9712_dapm_event(struct snd_soc_codec *codec, int event)
+static int wm9712_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
{
- switch (event) {
- case SNDRV_CTL_POWER_D0: /* full On */
- case SNDRV_CTL_POWER_D1: /* partial On */
- case SNDRV_CTL_POWER_D2: /* partial On */
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ case SND_SOC_BIAS_PREPARE:
break;
- case SNDRV_CTL_POWER_D3hot: /* Off, with power */
+ case SND_SOC_BIAS_STANDBY:
ac97_write(codec, AC97_POWERDOWN, 0x0000);
break;
- case SNDRV_CTL_POWER_D3cold: /* Off, without power */
+ case SND_SOC_BIAS_OFF:
/* disable everything including AC link */
ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff);
ac97_write(codec, AC97_POWERDOWN, 0xffff);
break;
}
- codec->dapm_state = event;
+ codec->bias_level = level;
return 0;
}
@@ -598,12 +589,12 @@ static int wm9712_reset(struct snd_soc_codec *codec, int try_warm)
{
if (try_warm && soc_ac97_ops.warm_reset) {
soc_ac97_ops.warm_reset(codec->ac97);
- if (!(ac97_read(codec, 0) & 0x8000))
+ if (ac97_read(codec, 0) == wm9712_reg[0])
return 1;
}
soc_ac97_ops.reset(codec->ac97);
- if (ac97_read(codec, 0) & 0x8000)
+ if (ac97_read(codec, 0) != wm9712_reg[0])
goto err;
return 0;
@@ -618,7 +609,7 @@ static int wm9712_soc_suspend(struct platform_device *pdev,
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->codec;
- wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+ wm9712_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
@@ -635,7 +626,7 @@ static int wm9712_soc_resume(struct platform_device *pdev)
return ret;
}
- wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+ wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
if (ret == 0) {
/* Sync reg_cache with the hardware after cold reset */
@@ -647,8 +638,8 @@ static int wm9712_soc_resume(struct platform_device *pdev)
}
}
- if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0)
- wm9712_dapm_event(codec, SNDRV_CTL_POWER_D0);
+ if (codec->suspend_bias_level == SND_SOC_BIAS_ON)
+ wm9712_set_bias_level(codec, SND_SOC_BIAS_ON);
return ret;
}
@@ -682,7 +673,7 @@ static int wm9712_soc_probe(struct platform_device *pdev)
codec->num_dai = ARRAY_SIZE(wm9712_dai);
codec->write = ac97_write;
codec->read = ac97_read;
- codec->dapm_event = wm9712_dapm_event;
+ codec->set_bias_level = wm9712_set_bias_level;
INIT_LIST_HEAD(&codec->dapm_widgets);
INIT_LIST_HEAD(&codec->dapm_paths);
@@ -706,7 +697,7 @@ static int wm9712_soc_probe(struct platform_device *pdev)
/* set alc mux to none */
ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000);
- wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+ wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
wm9712_add_controls(codec);
wm9712_add_widgets(codec);
ret = snd_soc_register_card(socdev);
diff --git a/sound/soc/codecs/wm9712.h b/sound/soc/codecs/wm9712.h
index 719105d61e6..d29e8a18ca6 100644
--- a/sound/soc/codecs/wm9712.h
+++ b/sound/soc/codecs/wm9712.h
@@ -8,7 +8,7 @@
#define WM9712_DAI_AC97_HIFI 0
#define WM9712_DAI_AC97_AUX 1
-extern struct snd_soc_codec_dai wm9712_dai[2];
+extern struct snd_soc_dai wm9712_dai[2];
extern struct snd_soc_codec_device soc_codec_dev_wm9712;
#endif
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 1f241161445..38d1fe0971f 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -10,9 +10,6 @@
* Free Software Foundation; either version 2 of the License, or (at your
* option) any later version.
*
- * Revision history
- * 4th Feb 2006 Initial version.
- *
* Features:-
*
* o Support for AC97 Codec, Voice DAC and Aux DAC
@@ -456,7 +453,7 @@ SND_SOC_DAPM_INPUT("MIC2B"),
SND_SOC_DAPM_VMID("VMID"),
};
-static const char *audio_map[][3] = {
+static const struct snd_soc_dapm_route audio_map[] = {
/* left HP mixer */
{"Left HP Mixer", "PC Beep Playback Switch", "PCBEEP"},
{"Left HP Mixer", "Voice Playback Switch", "Voice DAC"},
@@ -607,21 +604,14 @@ static const char *audio_map[][3] = {
{"Capture Mono Mux", "Stereo", "Capture Mixer"},
{"Capture Mono Mux", "Left", "Left Capture Source"},
{"Capture Mono Mux", "Right", "Right Capture Source"},
-
- {NULL, NULL, NULL},
};
static int wm9713_add_widgets(struct snd_soc_codec *codec)
{
- int i;
-
- for (i = 0; i < ARRAY_SIZE(wm9713_dapm_widgets); i++)
- snd_soc_dapm_new_control(codec, &wm9713_dapm_widgets[i]);
+ snd_soc_dapm_new_controls(codec, wm9713_dapm_widgets,
+ ARRAY_SIZE(wm9713_dapm_widgets));
- /* set up audio path audio_mapnects */
- for (i = 0; audio_map[i][0] != NULL; i++)
- snd_soc_dapm_connect_input(codec, audio_map[i][0],
- audio_map[i][1], audio_map[i][2]);
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
snd_soc_dapm_new_widgets(codec);
return 0;
@@ -799,7 +789,7 @@ static int wm9713_set_pll(struct snd_soc_codec *codec,
return 0;
}
-static int wm9713_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
+static int wm9713_set_dai_pll(struct snd_soc_dai *codec_dai,
int pll_id, unsigned int freq_in, unsigned int freq_out)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -810,7 +800,7 @@ static int wm9713_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
* Tristate the PCM DAI lines, tristate can be disabled by calling
* wm9713_set_dai_fmt()
*/
-static int wm9713_set_dai_tristate(struct snd_soc_codec_dai *codec_dai,
+static int wm9713_set_dai_tristate(struct snd_soc_dai *codec_dai,
int tristate)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -826,7 +816,7 @@ static int wm9713_set_dai_tristate(struct snd_soc_codec_dai *codec_dai,
* Configure WM9713 clock dividers.
* Voice DAC needs 256 FS
*/
-static int wm9713_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
+static int wm9713_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
int div_id, int div)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -868,7 +858,7 @@ static int wm9713_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
return 0;
}
-static int wm9713_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm9713_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -886,7 +876,7 @@ static int wm9713_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
gpio |= 0x0018;
break;
case SND_SOC_DAIFMT_CBS_CFS:
- reg |= 0x0200;
+ reg |= 0x2000;
gpio |= 0x001a;
break;
case SND_SOC_DAIFMT_CBS_CFM:
@@ -1011,15 +1001,24 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream)
return ac97_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate);
}
-#define WM9713_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
- SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\
- SNDRV_PCM_RATE_48000)
+#define WM9713_RATES (SNDRV_PCM_RATE_8000 | \
+ SNDRV_PCM_RATE_11025 | \
+ SNDRV_PCM_RATE_22050 | \
+ SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000)
+
+#define WM9713_PCM_RATES (SNDRV_PCM_RATE_8000 | \
+ SNDRV_PCM_RATE_11025 | \
+ SNDRV_PCM_RATE_16000 | \
+ SNDRV_PCM_RATE_22050 | \
+ SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000)
#define WM9713_PCM_FORMATS \
(SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \
SNDRV_PCM_FORMAT_S24_LE)
-struct snd_soc_codec_dai wm9713_dai[] = {
+struct snd_soc_dai wm9713_dai[] = {
{
.name = "AC97 HiFi",
.type = SND_SOC_DAI_AC97_BUS,
@@ -1061,13 +1060,13 @@ struct snd_soc_codec_dai wm9713_dai[] = {
.stream_name = "Voice Playback",
.channels_min = 1,
.channels_max = 1,
- .rates = WM9713_RATES,
+ .rates = WM9713_PCM_RATES,
.formats = WM9713_PCM_FORMATS,},
.capture = {
.stream_name = "Voice Capture",
.channels_min = 1,
.channels_max = 2,
- .rates = WM9713_RATES,
+ .rates = WM9713_PCM_RATES,
.formats = WM9713_PCM_FORMATS,},
.ops = {
.hw_params = wm9713_pcm_hw_params,
@@ -1086,44 +1085,44 @@ int wm9713_reset(struct snd_soc_codec *codec, int try_warm)
{
if (try_warm && soc_ac97_ops.warm_reset) {
soc_ac97_ops.warm_reset(codec->ac97);
- if (!(ac97_read(codec, 0) & 0x8000))
+ if (ac97_read(codec, 0) == wm9713_reg[0])
return 1;
}
soc_ac97_ops.reset(codec->ac97);
- if (ac97_read(codec, 0) & 0x8000)
+ if (ac97_read(codec, 0) != wm9713_reg[0])
return -EIO;
return 0;
}
EXPORT_SYMBOL_GPL(wm9713_reset);
-static int wm9713_dapm_event(struct snd_soc_codec *codec, int event)
+static int wm9713_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
{
u16 reg;
- switch (event) {
- case SNDRV_CTL_POWER_D0: /* full On */
+ switch (level) {
+ case SND_SOC_BIAS_ON:
/* enable thermal shutdown */
reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x1bff;
ac97_write(codec, AC97_EXTENDED_MID, reg);
break;
- case SNDRV_CTL_POWER_D1: /* partial On */
- case SNDRV_CTL_POWER_D2: /* partial On */
+ case SND_SOC_BIAS_PREPARE:
break;
- case SNDRV_CTL_POWER_D3hot: /* Off, with power */
+ case SND_SOC_BIAS_STANDBY:
/* enable master bias and vmid */
reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x3bff;
ac97_write(codec, AC97_EXTENDED_MID, reg);
ac97_write(codec, AC97_POWERDOWN, 0x0000);
break;
- case SNDRV_CTL_POWER_D3cold: /* Off, without power */
+ case SND_SOC_BIAS_OFF:
/* disable everything including AC link */
ac97_write(codec, AC97_EXTENDED_MID, 0xffff);
ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff);
ac97_write(codec, AC97_POWERDOWN, 0xffff);
break;
}
- codec->dapm_state = event;
+ codec->bias_level = level;
return 0;
}
@@ -1160,7 +1159,7 @@ static int wm9713_soc_resume(struct platform_device *pdev)
return ret;
}
- wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+ wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
/* do we need to re-start the PLL ? */
if (wm9713->pll_out)
@@ -1176,8 +1175,8 @@ static int wm9713_soc_resume(struct platform_device *pdev)
}
}
- if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0)
- wm9713_dapm_event(codec, SNDRV_CTL_POWER_D0);
+ if (codec->suspend_bias_level == SND_SOC_BIAS_ON)
+ wm9713_set_bias_level(codec, SND_SOC_BIAS_ON);
return ret;
}
@@ -1216,7 +1215,7 @@ static int wm9713_soc_probe(struct platform_device *pdev)
codec->num_dai = ARRAY_SIZE(wm9713_dai);
codec->write = ac97_write;
codec->read = ac97_read;
- codec->dapm_event = wm9713_dapm_event;
+ codec->set_bias_level = wm9713_set_bias_level;
INIT_LIST_HEAD(&codec->dapm_widgets);
INIT_LIST_HEAD(&codec->dapm_paths);
@@ -1238,7 +1237,7 @@ static int wm9713_soc_probe(struct platform_device *pdev)
goto reset_err;
}
- wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+ wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
/* unmute the adc - move to kcontrol */
reg = ac97_read(codec, AC97_CD) & 0x7fff;
diff --git a/sound/soc/codecs/wm9713.h b/sound/soc/codecs/wm9713.h
index d357b6c8134..63b8d81756e 100644
--- a/sound/soc/codecs/wm9713.h
+++ b/sound/soc/codecs/wm9713.h
@@ -46,7 +46,7 @@
#define WM9713_DAI_PCM_VOICE 2
extern struct snd_soc_codec_device soc_codec_dev_wm9713;
-extern struct snd_soc_codec_dai wm9713_dai[3];
+extern struct snd_soc_dai wm9713_dai[3];
int wm9713_reset(struct snd_soc_codec *codec, int try_warm);
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-01 18:23:21 +0000