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authorChristian Pellegrin <chripell@gmail.com>2008-11-15 08:58:16 +0100
committerMark Brown <broonie@opensource.wolfsonmicro.com>2008-11-17 11:45:39 +0000
commit1cad1de1b216b355a60d907c103b2daf1a285345 (patch)
tree737b36d3d003a0e18010f5be3d2c90ed510ba591 /sound/soc
parent6e5d9db271ab57789b09bcc61083ab71b7eabea9 (diff)
downloadtalos-obmc-linux-1cad1de1b216b355a60d907c103b2daf1a285345.tar.gz
talos-obmc-linux-1cad1de1b216b355a60d907c103b2daf1a285345.zip
ASoC: UDA134x codec driver
Signed-off-by: Christian Pellegrin <chripell@fsfe.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Diffstat (limited to 'sound/soc')
-rw-r--r--sound/soc/codecs/Kconfig8
-rw-r--r--sound/soc/codecs/Makefile4
-rw-r--r--sound/soc/codecs/l3.c91
-rw-r--r--sound/soc/codecs/uda134x.c656
-rw-r--r--sound/soc/codecs/uda134x_codec.h36
5 files changed, 795 insertions, 0 deletions
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 8a84460a6f74..04f49f5c3c3d 100644
--- a/sound/soc/codecs/Kconfig
+++ b/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
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 7ae17a6ea271..de6572356d1b 100644
--- a/sound/soc/codecs/Makefile
+++ b/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
diff --git a/sound/soc/codecs/l3.c b/sound/soc/codecs/l3.c
new file mode 100644
index 000000000000..5353af58862c
--- /dev/null
+++ b/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");
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
new file mode 100644
index 000000000000..04b30da10228
--- /dev/null
+++ b/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");
diff --git a/sound/soc/codecs/uda134x_codec.h b/sound/soc/codecs/uda134x_codec.h
new file mode 100644
index 000000000000..94f440490b31
--- /dev/null
+++ b/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
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