ASoC: UDA134x codec driver

Signed-off-by: Christian Pellegrin <chripell@fsfe.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

include/sound/l3.h

@@ -0,0 +1,18 @@
+#ifndef _L3_H_
+#define _L3_H_ 1
+
+struct l3_pins {
+	void (*setdat)(int);
+	void (*setclk)(int);
+	void (*setmode)(int);
+	int data_hold;
+	int data_setup;
+	int clock_high;
+	int mode_hold;
+	int mode;
+	int mode_setup;
+};
+
+int l3_write(struct l3_pins *adap, u8 addr, u8 *data, int len);
+
+#endif

include/sound/uda134x.h

@@ -0,0 +1,26 @@
+/*
+ * uda134x.h  --  UDA134x ALSA SoC Codec driver
+ *
+ * Copyright 2007 Dension Audio Systems Ltd.
+ * Author: Zoltan Devai
+ *
+ * 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 _UDA134X_H
+#define _UDA134X_H
+
+#include <sound/l3.h>
+
+struct uda134x_platform_data {
+	struct l3_pins l3;
+	void (*power) (int);
+	int model;
+#define UDA134X_UDA1340 1
+#define UDA134X_UDA1341 2
+#define UDA134X_UDA1344 3
+};
+
+#endif /* _UDA134X_H */

sound/soc/codecs/Kconfig

@@ -10,6 +10,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_TLV320AIC26 if SPI_MASTER
 	select SND_SOC_TLV320AIC3X if I2C
 	select SND_SOC_TWL4030 if TWL4030_CORE
+	select SND_SOC_UDA134X
 	select SND_SOC_UDA1380 if I2C
 	select SND_SOC_WM8510 if (I2C || SPI_MASTER)
 	select SND_SOC_WM8580 if I2C
@@ -66,6 +67,9 @@ config SND_SOC_CS4270_VD33_ERRATA
 	bool
 	depends on SND_SOC_CS4270
 
+config SND_SOC_L3
+       tristate
+
 config SND_SOC_SSM2602
 	tristate
 
@@ -85,6 +89,10 @@ config SND_SOC_TWL4030
 	tristate
 	depends on TWL4030_CORE
 
+config SND_SOC_UDA134X
+       tristate
+       select SND_SOC_L3
+
 config SND_SOC_UDA1380
         tristate
 

sound/soc/codecs/Makefile

@@ -3,11 +3,13 @@ snd-soc-ad1980-objs := ad1980.o
 snd-soc-ad73311-objs := ad73311.o
 snd-soc-ak4535-objs := ak4535.o
 snd-soc-cs4270-objs := cs4270.o
+snd-soc-l3-objs := l3.o
 snd-soc-ssm2602-objs := ssm2602.o
 snd-soc-tlv320aic23-objs := tlv320aic23.o
 snd-soc-tlv320aic26-objs := tlv320aic26.o
 snd-soc-tlv320aic3x-objs := tlv320aic3x.o
 snd-soc-twl4030-objs := twl4030.o
+snd-soc-uda134x-objs := uda134x.o
 snd-soc-uda1380-objs := uda1380.o
 snd-soc-wm8510-objs := wm8510.o
 snd-soc-wm8580-objs := wm8580.o
@@ -27,11 +29,13 @@ obj-$(CONFIG_SND_SOC_AD1980)	+= snd-soc-ad1980.o
 obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
 obj-$(CONFIG_SND_SOC_AK4535)	+= snd-soc-ak4535.o
 obj-$(CONFIG_SND_SOC_CS4270)	+= snd-soc-cs4270.o
+obj-$(CONFIG_SND_SOC_L3)	+= snd-soc-l3.o
 obj-$(CONFIG_SND_SOC_SSM2602)	+= snd-soc-ssm2602.o
 obj-$(CONFIG_SND_SOC_TLV320AIC23)	+= snd-soc-tlv320aic23.o
 obj-$(CONFIG_SND_SOC_TLV320AIC26)	+= snd-soc-tlv320aic26.o
 obj-$(CONFIG_SND_SOC_TLV320AIC3X)	+= snd-soc-tlv320aic3x.o
 obj-$(CONFIG_SND_SOC_TWL4030)	+= snd-soc-twl4030.o
+obj-$(CONFIG_SND_SOC_UDA134X)	+= snd-soc-uda134x.o
 obj-$(CONFIG_SND_SOC_UDA1380)	+= snd-soc-uda1380.o
 obj-$(CONFIG_SND_SOC_WM8510)	+= snd-soc-wm8510.o
 obj-$(CONFIG_SND_SOC_WM8580)	+= snd-soc-wm8580.o

sound/soc/codecs/l3.c

@@ -0,0 +1,91 @@
+/*
+ * L3 code
+ *
+ *  Copyright (C) 2008, Christian Pellegrin <chripell@evolware.org>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ *
+ * based on:
+ *
+ * L3 bus algorithm module.
+ *
+ *  Copyright (C) 2001 Russell King, All Rights Reserved.
+ *
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/kernel.h>
+#include <linux/delay.h>
+
+#include <sound/l3.h>
+
+/*
+ * Send one byte of data to the chip.  Data is latched into the chip on
+ * the rising edge of the clock.
+ */
+static void sendbyte(struct l3_pins *adap, unsigned int byte)
+{
+	int i;
+
+	for (i = 0; i < 8; i++) {
+		adap->setclk(0);
+		udelay(adap->data_hold);
+		adap->setdat(byte & 1);
+		udelay(adap->data_setup);
+		adap->setclk(1);
+		udelay(adap->clock_high);
+		byte >>= 1;
+	}
+}
+
+/*
+ * Send a set of bytes to the chip.  We need to pulse the MODE line
+ * between each byte, but never at the start nor at the end of the
+ * transfer.
+ */
+static void sendbytes(struct l3_pins *adap, const u8 *buf,
+		      int len)
+{
+	int i;
+
+	for (i = 0; i < len; i++) {
+		if (i) {
+			udelay(adap->mode_hold);
+			adap->setmode(0);
+			udelay(adap->mode);
+		}
+		adap->setmode(1);
+		udelay(adap->mode_setup);
+		sendbyte(adap, buf[i]);
+	}
+}
+
+int l3_write(struct l3_pins *adap, u8 addr, u8 *data, int len)
+{
+	adap->setclk(1);
+	adap->setdat(1);
+	adap->setmode(1);
+	udelay(adap->mode);
+
+	adap->setmode(0);
+	udelay(adap->mode_setup);
+	sendbyte(adap, addr);
+	udelay(adap->mode_hold);
+
+	sendbytes(adap, data, len);
+
+	adap->setclk(1);
+	adap->setdat(1);
+	adap->setmode(0);
+
+	return len;
+}
+EXPORT_SYMBOL_GPL(l3_write);
+
+MODULE_DESCRIPTION("L3 bit-banging driver");
+MODULE_AUTHOR("Christian Pellegrin <chripell@evolware.org>");
+MODULE_LICENSE("GPL");

sound/soc/codecs/uda134x.c

@@ -0,0 +1,656 @@
+/*
+ * uda134x.c  --  UDA134X ALSA SoC Codec driver
+ *
+ * Modifications by Christian Pellegrin <chripell@evolware.org>
+ *
+ * Copyright 2007 Dension Audio Systems Ltd.
+ * Author: Zoltan Devai
+ *
+ * Based on the WM87xx drivers by Liam Girdwood and Richard Purdie
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/delay.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include <sound/uda134x.h>
+#include <sound/l3.h>
+
+#include "uda134x_codec.h"
+
+
+#define POWER_OFF_ON_STANDBY 1
+/*
+  ALSA SOC usually puts the device in standby mode when it's not used
+  for sometime. If you define POWER_OFF_ON_STANDBY the driver will
+  turn off the ADC/DAC when this callback is invoked and turn it back
+  on when needed. Unfortunately this will result in a very light bump
+  (it can be audible only with good earphones). If this bothers you
+  just comment this line, you will have slightly higher power
+  consumption . Please note that sending the L3 command for ADC is
+  enough to make the bump, so it doesn't make difference if you
+  completely take off power from the codec.
+ */
+
+#define UDA134X_RATES SNDRV_PCM_RATE_8000_48000
+#define UDA134X_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \
+		SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE)
+
+struct uda134x_priv {
+	int sysclk;
+	int dai_fmt;
+
+	struct snd_pcm_substream *master_substream;
+	struct snd_pcm_substream *slave_substream;
+};
+
+/* In-data addresses are hard-coded into the reg-cache values */
+static const char uda134x_reg[UDA134X_REGS_NUM] = {
+	/* Extended address registers */
+	0x04, 0x04, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00,
+	/* Status, data regs */
+	0x00, 0x83, 0x00, 0x40, 0x80, 0x00,
+};
+
+/*
+ * The codec has no support for reading its registers except for peak level...
+ */
+static inline unsigned int uda134x_read_reg_cache(struct snd_soc_codec *codec,
+	unsigned int reg)
+{
+	u8 *cache = codec->reg_cache;
+
+	if (reg >= UDA134X_REGS_NUM)
+		return -1;
+	return cache[reg];
+}
+
+/*
+ * Write the register cache
+ */
+static inline void uda134x_write_reg_cache(struct snd_soc_codec *codec,
+	u8 reg, unsigned int value)
+{
+	u8 *cache = codec->reg_cache;
+
+	if (reg >= UDA134X_REGS_NUM)
+		return;
+	cache[reg] = value;
+}
+
+/*
+ * Write to the uda134x registers
+ *
+ */
+static int uda134x_write(struct snd_soc_codec *codec, unsigned int reg,
+	unsigned int value)
+{
+	int ret;
+	u8 addr;
+	u8 data = value;
+	struct uda134x_platform_data *pd = codec->control_data;
+
+	pr_debug("%s reg: %02X, value:%02X\n", __func__, reg, value);
+
+	if (reg >= UDA134X_REGS_NUM) {
+		printk(KERN_ERR "%s unkown register: reg: %d",
+		       __func__, reg);
+		return -EINVAL;
+	}
+
+	uda134x_write_reg_cache(codec, reg, value);
+
+	switch (reg) {
+	case UDA134X_STATUS0:
+	case UDA134X_STATUS1:
+		addr = UDA134X_STATUS_ADDR;
+		break;
+	case UDA134X_DATA000:
+	case UDA134X_DATA001:
+	case UDA134X_DATA010:
+		addr = UDA134X_DATA0_ADDR;
+		break;
+	case UDA134X_DATA1:
+		addr = UDA134X_DATA1_ADDR;
+		break;
+	default:
+		/* It's an extended address register */
+		addr =  (reg | UDA134X_EXTADDR_PREFIX);
+
+		ret = l3_write(&pd->l3,
+			       UDA134X_DATA0_ADDR, &addr, 1);
+		if (ret != 1)
+			return -EIO;
+
+		addr = UDA134X_DATA0_ADDR;
+		data = (value | UDA134X_EXTDATA_PREFIX);
+		break;
+	}
+
+	ret = l3_write(&pd->l3,
+		       addr, &data, 1);
+	if (ret != 1)
+		return -EIO;
+
+	return 0;
+}
+
+static inline void uda134x_reset(struct snd_soc_codec *codec)
+{
+	u8 reset_reg = uda134x_read_reg_cache(codec, UDA134X_STATUS0);
+	uda134x_write(codec, UDA134X_STATUS0, reset_reg | (1<<6));
+	msleep(1);
+	uda134x_write(codec, UDA134X_STATUS0, reset_reg & ~(1<<6));
+}
+
+static int uda134x_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u8 mute_reg = uda134x_read_reg_cache(codec, UDA134X_DATA010);
+
+	pr_debug("%s mute: %d\n", __func__, mute);
+
+	if (mute)
+		mute_reg |= (1<<2);
+	else
+		mute_reg &= ~(1<<2);
+
+	uda134x_write(codec, UDA134X_DATA010, mute_reg & ~(1<<2));
+
+	return 0;
+}
+
+static int uda134x_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	struct uda134x_priv *uda134x = codec->private_data;
+	struct snd_pcm_runtime *master_runtime;
+
+	if (uda134x->master_substream) {
+		master_runtime = uda134x->master_substream->runtime;
+
+		pr_debug("%s constraining to %d bits at %d\n", __func__,
+			 master_runtime->sample_bits,
+			 master_runtime->rate);
+
+		snd_pcm_hw_constraint_minmax(substream->runtime,
+					     SNDRV_PCM_HW_PARAM_RATE,
+					     master_runtime->rate,
+					     master_runtime->rate);
+
+		snd_pcm_hw_constraint_minmax(substream->runtime,
+					     SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+					     master_runtime->sample_bits,
+					     master_runtime->sample_bits);
+
+		uda134x->slave_substream = substream;
+	} else
+		uda134x->master_substream = substream;
+
+	return 0;
+}
+
+static void uda134x_shutdown(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	struct uda134x_priv *uda134x = codec->private_data;
+
+	if (uda134x->master_substream == substream)
+		uda134x->master_substream = uda134x->slave_substream;
+
+	uda134x->slave_substream = NULL;
+}
+
+static int uda134x_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	struct uda134x_priv *uda134x = codec->private_data;
+	u8 hw_params;
+
+	if (substream == uda134x->slave_substream) {
+		pr_debug("%s ignoring hw_params for slave substream\n",
+			 __func__);
+		return 0;
+	}
+
+	hw_params = uda134x_read_reg_cache(codec, UDA134X_STATUS0);
+	hw_params &= STATUS0_SYSCLK_MASK;
+	hw_params &= STATUS0_DAIFMT_MASK;
+
+	pr_debug("%s sysclk: %d, rate:%d\n", __func__,
+		 uda134x->sysclk, params_rate(params));
+
+	/* set SYSCLK / fs ratio */
+	switch (uda134x->sysclk / params_rate(params)) {
+	case 512:
+		break;
+	case 384:
+		hw_params |= (1<<4);
+		break;
+	case 256:
+		hw_params |= (1<<5);
+		break;
+	default:
+		printk(KERN_ERR "%s unsupported fs\n", __func__);
+		return -EINVAL;
+	}
+
+	pr_debug("%s dai_fmt: %d, params_format:%d\n", __func__,
+		 uda134x->dai_fmt, params_format(params));
+
+	/* set DAI format and word length */
+	switch (uda134x->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		switch (params_format(params)) {
+		case SNDRV_PCM_FORMAT_S16_LE:
+			hw_params |= (1<<1);
+			break;
+		case SNDRV_PCM_FORMAT_S18_3LE:
+			hw_params |= (1<<2);
+			break;
+		case SNDRV_PCM_FORMAT_S20_3LE:
+			hw_params |= ((1<<2) | (1<<1));
+			break;
+		default:
+			printk(KERN_ERR "%s unsupported format (right)\n",
+			       __func__);
+			return -EINVAL;
+		}
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		hw_params |= (1<<3);
+		break;
+	default:
+		printk(KERN_ERR "%s unsupported format\n", __func__);
+		return -EINVAL;
+	}
+
+	uda134x_write(codec, UDA134X_STATUS0, hw_params);
+
+	return 0;
+}
+
+static int uda134x_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+				  int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct uda134x_priv *uda134x = codec->private_data;
+
+	pr_debug("%s clk_id: %d, freq: %d, dir: %d\n", __func__,
+		 clk_id, freq, dir);
+
+	/* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable
+	   because the codec is slave. Of course limitations of the clock
+	   master (the IIS controller) apply.
+	   We'll error out on set_hw_params if it's not OK */
+	if ((freq >= (256 * 8000)) && (freq <= (512 * 48000))) {
+		uda134x->sysclk = freq;
+		return 0;
+	}
+
+	printk(KERN_ERR "%s unsupported sysclk\n", __func__);
+	return -EINVAL;
+}
+
+static int uda134x_set_dai_fmt(struct snd_soc_dai *codec_dai,
+			       unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct uda134x_priv *uda134x = codec->private_data;
+
+	pr_debug("%s fmt: %08X\n", __func__, fmt);
+
+	/* codec supports only full slave mode */
+	if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) {
+		printk(KERN_ERR "%s unsupported slave mode\n", __func__);
+		return -EINVAL;
+	}
+
+	/* no support for clock inversion */
+	if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) {
+		printk(KERN_ERR "%s unsupported clock inversion\n", __func__);
+		return -EINVAL;
+	}
+
+	/* We can't setup DAI format here as it depends on the word bit num */
+	/* so let's just store the value for later */
+	uda134x->dai_fmt = fmt;
+
+	return 0;
+}
+
+static int uda134x_set_bias_level(struct snd_soc_codec *codec,
+				  enum snd_soc_bias_level level)
+{
+	u8 reg;
+	struct uda134x_platform_data *pd = codec->control_data;
+	int i;
+	u8 *cache = codec->reg_cache;
+
+	pr_debug("%s bias level %d\n", __func__, level);
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		/* ADC, DAC on */
+		reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1);
+		uda134x_write(codec, UDA134X_STATUS1, reg | 0x03);
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		/* power on */
+		if (pd->power) {
+			pd->power(1);
+			/* Sync reg_cache with the hardware */
+			for (i = 0; i < ARRAY_SIZE(uda134x_reg); i++)
+				codec->write(codec, i, *cache++);
+		}
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		/* ADC, DAC power off */
+		reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1);
+		uda134x_write(codec, UDA134X_STATUS1, reg & ~(0x03));
+		break;
+	case SND_SOC_BIAS_OFF:
+		/* power off */
+		if (pd->power)
+			pd->power(0);
+		break;
+	}
+	codec->bias_level = level;
+	return 0;
+}
+
+static const char *uda134x_dsp_setting[] = {"Flat", "Minimum1",
+					    "Minimum2", "Maximum"};
+static const char *uda134x_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
+static const char *uda134x_mixmode[] = {"Differential", "Analog1",
+					"Analog2", "Both"};
+
+static const struct soc_enum uda134x_mixer_enum[] = {
+SOC_ENUM_SINGLE(UDA134X_DATA010, 0, 0x04, uda134x_dsp_setting),
+SOC_ENUM_SINGLE(UDA134X_DATA010, 3, 0x04, uda134x_deemph),
+SOC_ENUM_SINGLE(UDA134X_EA010, 0, 0x04, uda134x_mixmode),
+};
+
+static const struct snd_kcontrol_new uda1341_snd_controls[] = {
+SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1),
+SOC_SINGLE("Capture Volume", UDA134X_EA010, 2, 0x07, 0),
+SOC_SINGLE("Analog1 Volume", UDA134X_EA000, 0, 0x1F, 1),
+SOC_SINGLE("Analog2 Volume", UDA134X_EA001, 0, 0x1F, 1),
+
+SOC_SINGLE("Mic Sensitivity", UDA134X_EA010, 2, 7, 0),
+SOC_SINGLE("Mic Volume", UDA134X_EA101, 0, 0x1F, 0),
+
+SOC_SINGLE("Tone Control - Bass", UDA134X_DATA001, 2, 0xF, 0),
+SOC_SINGLE("Tone Control - Treble", UDA134X_DATA001, 0, 3, 0),
+
+SOC_ENUM("Sound Processing Filter", uda134x_mixer_enum[0]),
+SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]),
+SOC_ENUM("Input Mux", uda134x_mixer_enum[2]),
+
+SOC_SINGLE("AGC Switch", UDA134X_EA100, 4, 1, 0),
+SOC_SINGLE("AGC Target Volume", UDA134X_EA110, 0, 0x03, 1),
+SOC_SINGLE("AGC Timing", UDA134X_EA110, 2, 0x07, 0),
+
+SOC_SINGLE("DAC +6dB Switch", UDA134X_STATUS1, 6, 1, 0),
+SOC_SINGLE("ADC +6dB Switch", UDA134X_STATUS1, 5, 1, 0),
+SOC_SINGLE("ADC Polarity Switch", UDA134X_STATUS1, 4, 1, 0),
+SOC_SINGLE("DAC Polarity Switch", UDA134X_STATUS1, 3, 1, 0),
+SOC_SINGLE("Double Speed Playback Switch", UDA134X_STATUS1, 2, 1, 0),
+SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new uda1340_snd_controls[] = {
+SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1),
+
+SOC_SINGLE("Tone Control - Bass", UDA134X_DATA001, 2, 0xF, 0),
+SOC_SINGLE("Tone Control - Treble", UDA134X_DATA001, 0, 3, 0),
+
+SOC_ENUM("Sound Processing Filter", uda134x_mixer_enum[0]),
+SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]),
+
+SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0),
+};
+
+static int uda134x_add_controls(struct snd_soc_codec *codec)
+{
+	int err, i, n;
+	const struct snd_kcontrol_new *ctrls;
+	struct uda134x_platform_data *pd = codec->control_data;
+
+	switch (pd->model) {
+	case UDA134X_UDA1340:
+	case UDA134X_UDA1344:
+		n = ARRAY_SIZE(uda1340_snd_controls);
+		ctrls = uda1340_snd_controls;
+		break;
+	case UDA134X_UDA1341:
+		n = ARRAY_SIZE(uda1341_snd_controls);
+		ctrls = uda1341_snd_controls;
+		break;
+	default:
+		printk(KERN_ERR "%s unkown codec type: %d",
+		       __func__, pd->model);
+		return -EINVAL;
+	}
+
+	for (i = 0; i < n; i++) {
+		err = snd_ctl_add(codec->card,
+				  snd_soc_cnew(&ctrls[i],
+					       codec, NULL));
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+
+struct snd_soc_dai uda134x_dai = {
+	.name = "UDA134X",
+	/* playback capabilities */
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = UDA134X_RATES,
+		.formats = UDA134X_FORMATS,
+	},
+	/* capture capabilities */
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = UDA134X_RATES,
+		.formats = UDA134X_FORMATS,
+	},
+	/* pcm operations */
+	.ops = {
+		.startup = uda134x_startup,
+		.shutdown = uda134x_shutdown,
+		.hw_params = uda134x_hw_params,
+	},
+	/* DAI operations */
+	.dai_ops = {
+		.digital_mute = uda134x_mute,
+		.set_sysclk = uda134x_set_dai_sysclk,
+		.set_fmt = uda134x_set_dai_fmt,
+	}
+};
+EXPORT_SYMBOL(uda134x_dai);
+
+
+static int uda134x_soc_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	struct uda134x_priv *uda134x;
+	void *codec_setup_data = socdev->codec_data;
+	int ret = -ENOMEM;
+	struct uda134x_platform_data *pd;
+
+	printk(KERN_INFO "UDA134X SoC Audio Codec\n");
+
+	if (!codec_setup_data) {
+		printk(KERN_ERR "UDA134X SoC codec: "
+		       "missing L3 bitbang function\n");
+		return -ENODEV;
+	}
+
+	pd = codec_setup_data;
+	switch (pd->model) {
+	case UDA134X_UDA1340:
+	case UDA134X_UDA1341:
+	case UDA134X_UDA1344:
+		break;
+	default:
+		printk(KERN_ERR "UDA134X SoC codec: "
+		       "unsupported model %d\n",
+			pd->model);
+		return -EINVAL;
+	}
+
+	socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (socdev->codec == NULL)
+		return ret;
+
+	codec = socdev->codec;
+
+	uda134x = kzalloc(sizeof(struct uda134x_priv), GFP_KERNEL);
+	if (uda134x == NULL)
+		goto priv_err;
+	codec->private_data = uda134x;
+
+	codec->reg_cache = kmemdup(uda134x_reg, sizeof(uda134x_reg),
+				   GFP_KERNEL);
+	if (codec->reg_cache == NULL)
+		goto reg_err;
+
+	mutex_init(&codec->mutex);
+
+	codec->reg_cache_size = sizeof(uda134x_reg);
+	codec->reg_cache_step = 1;
+
+	codec->name = "UDA134X";
+	codec->owner = THIS_MODULE;
+	codec->dai = &uda134x_dai;
+	codec->num_dai = 1;
+	codec->read = uda134x_read_reg_cache;
+	codec->write = uda134x_write;
+#ifdef POWER_OFF_ON_STANDBY
+	codec->set_bias_level = uda134x_set_bias_level;
+#endif
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	codec->control_data = codec_setup_data;
+
+	if (pd->power)
+		pd->power(1);
+
+	uda134x_reset(codec);
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		printk(KERN_ERR "UDA134X: failed to register pcms\n");
+		goto pcm_err;
+	}
+
+	ret = uda134x_add_controls(codec);
+	if (ret < 0) {
+		printk(KERN_ERR "UDA134X: failed to register controls\n");
+		goto pcm_err;
+	}
+
+	ret = snd_soc_register_card(socdev);
+	if (ret < 0) {
+		printk(KERN_ERR "UDA134X: failed to register card\n");
+		goto card_err;
+	}
+
+	return 0;
+
+card_err:
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+pcm_err:
+	kfree(codec->reg_cache);
+reg_err:
+	kfree(codec->private_data);
+priv_err:
+	kfree(codec);
+	return ret;
+}
+
+/* power down chip */
+static int uda134x_soc_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+
+	kfree(codec->private_data);
+	kfree(codec->reg_cache);
+	kfree(codec);
+
+	return 0;
+}
+
+#if defined(CONFIG_PM)
+static int uda134x_soc_suspend(struct platform_device *pdev,
+						pm_message_t state)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static int uda134x_soc_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	uda134x_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
+	uda134x_set_bias_level(codec, SND_SOC_BIAS_ON);
+	return 0;
+}
+#else
+#define uda134x_soc_suspend NULL
+#define uda134x_soc_resume NULL
+#endif /* CONFIG_PM */
+
+struct snd_soc_codec_device soc_codec_dev_uda134x = {
+	.probe =        uda134x_soc_probe,
+	.remove =       uda134x_soc_remove,
+	.suspend =      uda134x_soc_suspend,
+	.resume =       uda134x_soc_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_uda134x);
+
+MODULE_DESCRIPTION("UDA134X ALSA soc codec driver");
+MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell@evolware.org>");
+MODULE_LICENSE("GPL");

sound/soc/codecs/uda134x_codec.h

@@ -0,0 +1,36 @@
+#ifndef _UDA134X_CODEC_H
+#define _UDA134X_CODEC_H
+
+#define UDA134X_L3ADDR	5
+#define UDA134X_DATA0_ADDR	((UDA134X_L3ADDR << 2) | 0)
+#define UDA134X_DATA1_ADDR	((UDA134X_L3ADDR << 2) | 1)
+#define UDA134X_STATUS_ADDR	((UDA134X_L3ADDR << 2) | 2)
+
+#define UDA134X_EXTADDR_PREFIX	0xC0
+#define UDA134X_EXTDATA_PREFIX	0xE0
+
+/* UDA134X registers */
+#define UDA134X_EA000	0
+#define UDA134X_EA001	1
+#define UDA134X_EA010	2
+#define UDA134X_EA011	3
+#define UDA134X_EA100	4
+#define UDA134X_EA101	5
+#define UDA134X_EA110	6
+#define UDA134X_EA111	7
+#define UDA134X_STATUS0 8
+#define UDA134X_STATUS1 9
+#define UDA134X_DATA000 10
+#define UDA134X_DATA001 11
+#define UDA134X_DATA010 12
+#define UDA134X_DATA1	13
+
+#define UDA134X_REGS_NUM 14
+
+#define STATUS0_DAIFMT_MASK (~(7<<1))
+#define STATUS0_SYSCLK_MASK (~(3<<4))
+
+extern struct snd_soc_dai uda134x_dai;
+extern struct snd_soc_codec_device soc_codec_dev_uda134x;
+
+#endif