[ALSA] ASoC: Add UDA1380 driver

The UDA1380 codec is used by the HTC Magician and a number of Samsung
reference boards.

This driver has had a long out of tree history, having originally been
written by Giorgio Padrin and converted to ASoC by Richard Purdie.
Since conversion to ASoC most of the work on the driver has been done by
Philipp Zabel with some review and updates for new APIs by Liam Girdwood
and Mark Brown.

Signed-off-by: Richard Purdie <rpurdie@rpsys.net>
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

4 files changed, 946 insertions, 0 deletions

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index d4a5fe4..7beefcc 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -2,6 +2,9 @@ config SND_SOC_AC97_CODEC
 	tristate
 	select SND_AC97_CODEC
 
+config SND_SOC_UDA1380
+        tristate
+
 config SND_SOC_WM8731
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 4e1314c..d5926a1 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -1,4 +1,5 @@
 snd-soc-ac97-objs := ac97.o
+snd-soc-uda1380-objs := uda1380.o
 snd-soc-wm8731-objs := wm8731.o
 snd-soc-wm8750-objs := wm8750.o
 snd-soc-wm8753-objs := wm8753.o
@@ -8,6 +9,7 @@ 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_UDA1380)	+= snd-soc-uda1380.o
 obj-$(CONFIG_SND_SOC_WM8731)	+= snd-soc-wm8731.o
 obj-$(CONFIG_SND_SOC_WM8750)	+= snd-soc-wm8750.o
 obj-$(CONFIG_SND_SOC_WM8753)	+= snd-soc-wm8753.o
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
new file mode 100644
index 0000000..cb50486
--- /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_codec_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_codec_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_codec_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 = sizeof(uda1380_reg);
+	codec->reg_cache_step = 2;
+	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 0000000..f9d885c8
--- /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_codec_dai uda1380_dai[3];
+extern struct snd_soc_codec_device soc_codec_dev_uda1380;
+
+#endif /* _UDA1380_H */