[alsa-devel] [PATCH] ALSA SOC driver for s3c24xx with uda1341
Christian Pellegrin
chripell at gmail.com
Sat Nov 8 08:43:55 CET 2008
This patch adds support for the UDA1341 codec and a sound card
definition for one of these UDAs connected to the s3c24xx. It is
*heavily* based on the one made by Zoltan Devai but:
1 since the UDA is the only use of the L3 protocol I can find I just
embedded a stripped-down L3 bit-banging algorithm from the original
RMK work. It is really small.
2 the driver has the possibility to specify the pins used by codec
via platform data so it can work on SMDK2410, SMDK2440 or any custom
design.
3 it tries to guess the right clock source and divider so it is not tied
to a particular crystal used.
4 it fixes some bugs.
Thank you for reviews/comments.
Generated on 20081108 against v2.6.27
Signed-off-by: Christian Pellegrin <chripell at fsfe.org>
---
include/sound/s3c24xx_uda1341.h | 41 ++
sound/soc/codecs/Kconfig | 3 +
sound/soc/codecs/Makefile | 1 +
sound/soc/codecs/uda1341/Makefile | 3 +
sound/soc/codecs/uda1341/l3.c | 106 +++++
sound/soc/codecs/uda1341/l3.h | 8 +
sound/soc/codecs/uda1341/uda1341.c | 537 ++++++++++++++++++++++
sound/soc/codecs/uda1341/uda1341.h | 47 ++
sound/soc/codecs/uda1341/uda1341_platform_data.h | 18 +
sound/soc/s3c24xx/Kconfig | 7 +-
sound/soc/s3c24xx/Makefile | 2 +
sound/soc/s3c24xx/s3c24xx_uda1341.c | 314 +++++++++++++
12 files changed, 1086 insertions(+), 1 deletions(-)
diff --git a/include/sound/s3c24xx_uda1341.h b/include/sound/s3c24xx_uda1341.h
new file mode 100644
index 0000000..bd80bd3
--- /dev/null
+++ b/include/sound/s3c24xx_uda1341.h
@@ -0,0 +1,41 @@
+#ifndef _S3C24XX_UDA1341_H_
+#define _S3C24XX_UDA1341_H_ 1
+
+/*
+
+Example usage (pins for SMDK2410). Put something like this in your
+machine file:
+
+...
+
+static struct s3c24xx_uda1341_platform_data s3c24xx_uda1341_data = {
+ .l3_clk = S3C2410_GPB4,
+ .l3_data = S3C2410_GPB3,
+ .l3_mode = S3C2410_GPB2,
+};
+
+static struct platform_device s3c24xx_uda1341 = {
+ .name = "s3c24xx_uda1341",
+ .dev = {
+ .platform_data = &s3c24xx_uda1341_data,
+ }
+};
+
+...
+
+static struct platform_device *smdk2410_devices[] __initdata = {
+...
+ &s3c24xx_uda1341,
+...
+};
+
+ */
+
+struct s3c24xx_uda1341_platform_data {
+ int l3_clk;
+ int l3_mode;
+ int l3_data;
+ void (*power) (int);
+};
+
+#endif
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 1db04a2..4b483b7 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -50,3 +50,6 @@ config SND_SOC_CS4270_VD33_ERRATA
config SND_SOC_TLV320AIC3X
tristate
depends on I2C
+
+config SND_SOC_UDA1341
+ tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index d7b97ab..cbace60 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -23,3 +23,4 @@ obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o
obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o
obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o
obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o
+obj-$(CONFIG_SND_SOC_UDA1341) += uda1341/
diff --git a/sound/soc/codecs/uda1341/Makefile b/sound/soc/codecs/uda1341/Makefile
new file mode 100644
index 0000000..f9869ba
--- /dev/null
+++ b/sound/soc/codecs/uda1341/Makefile
@@ -0,0 +1,3 @@
+snd-soc-uda1341-objs := l3.o uda1341.o
+
+obj-$(CONFIG_SND_SOC_UDA1341) += snd-soc-uda1341.o
\ No newline at end of file
diff --git a/sound/soc/codecs/uda1341/l3.c b/sound/soc/codecs/uda1341/l3.c
new file mode 100644
index 0000000..b65b352
--- /dev/null
+++ b/sound/soc/codecs/uda1341/l3.c
@@ -0,0 +1,106 @@
+/*
+ * L3 code
+ *
+ * Copyright (C) 2008, Christian Pellegrin <chripell at 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 <linux/slab.h>
+#include <linux/init.h>
+#include <linux/errno.h>
+#include <linux/sched.h>
+
+#include "l3.h"
+
+/*#define L3_DEBUG 1*/
+#ifdef L3_DEBUG
+#define DBG(format, arg...) \
+ printk(KERN_DEBUG "L3: %s:" format "\n" , __func__, ## arg)
+#else
+#define DBG(format, arg...) do {} while (0)
+#endif
+
+#define setdat(adap, val) (adap->setdat(adap, val))
+#define setclk(adap, val) (adap->setclk(adap, val))
+#define setmode(adap, val) (adap->setmode(adap, val))
+
+/*
+ * 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 uda1341_platform_data *adap, unsigned int byte)
+{
+ int i;
+
+ DBG("%02x", byte);
+
+ for (i = 0; i < 8; i++) {
+ setclk(adap, 0);
+ udelay(adap->data_hold);
+ setdat(adap, byte & 1);
+ udelay(adap->data_setup);
+ setclk(adap, 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 uda1341_platform_data *adap, const u8 *buf,
+ int len)
+{
+ int i;
+
+ for (i = 0; i < len; i++) {
+ if (i) {
+ udelay(adap->mode_hold);
+ setmode(adap, 0);
+ udelay(adap->mode);
+ }
+ setmode(adap, 1);
+ udelay(adap->mode_setup);
+ sendbyte(adap, buf[i]);
+ }
+}
+
+int l3_write(struct uda1341_platform_data *adap, u8 addr, u8 *data, int len)
+{
+ setclk(adap, 1);
+ setdat(adap, 1);
+ setmode(adap, 1);
+ udelay(adap->mode);
+
+ setmode(adap, 0);
+ udelay(adap->mode_setup);
+ sendbyte(adap, addr);
+ udelay(adap->mode_hold);
+
+ sendbytes(adap, data, len);
+
+ setclk(adap, 1);
+ setdat(adap, 1);
+ setmode(adap, 0);
+
+ return len;
+}
+
+
diff --git a/sound/soc/codecs/uda1341/l3.h b/sound/soc/codecs/uda1341/l3.h
new file mode 100644
index 0000000..2c31e0c
--- /dev/null
+++ b/sound/soc/codecs/uda1341/l3.h
@@ -0,0 +1,8 @@
+#ifndef _L3_H_
+#define _L3_H_ 1
+
+#include "uda1341_platform_data.h"
+
+int l3_write(struct uda1341_platform_data *adap, u8 addr, u8 *data, int len);
+
+#endif
diff --git a/sound/soc/codecs/uda1341/uda1341.c b/sound/soc/codecs/uda1341/uda1341.c
new file mode 100644
index 0000000..93cadf0
--- /dev/null
+++ b/sound/soc/codecs/uda1341/uda1341.c
@@ -0,0 +1,537 @@
+/*
+ * uda1341.c -- UDA1341 ALSA SoC Codec driver
+ *
+ * Modifications by Christian Pellegrin <chripell at 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 "uda1341.h"
+#include "l3.h"
+
+/*#define UDA1341_DEBUG 1*/
+#ifdef UDA1341_DEBUG
+#define DBG(format, arg...) \
+ printk(KERN_DEBUG "UDA1341: %s:" format "\n" , __func__, ## arg)
+#else
+#define DBG(format, arg...) do {} while (0)
+#endif
+
+#define UDA1341_RATES SNDRV_PCM_RATE_8000_48000
+#define UDA1341_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE)
+
+struct uda1341_priv {
+ int sysclk;
+ int dai_fmt;
+};
+
+/* In-data addresses are hard-coded into the reg-cache values */
+static const char uda1341_reg[UDA1341_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 uda1341_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u8 *cache = codec->reg_cache;
+
+ if (reg >= UDA1341_REGS_NUM)
+ return -1;
+ return cache[reg];
+}
+
+/*
+ * Write the register cache
+ */
+static inline void uda1341_write_reg_cache(struct snd_soc_codec *codec,
+ u8 reg, unsigned int value)
+{
+ u8 *cache = codec->reg_cache;
+
+ if (reg >= UDA1341_REGS_NUM)
+ return;
+ cache[reg] = value;
+}
+
+/*
+ * Write to the uda1341 registers
+ *
+ */
+static int uda1341_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ int ret;
+ u8 addr;
+ u8 data = value;
+
+ DBG("reg: %02X, value:%02X", reg, value);
+
+ if (reg >= UDA1341_REGS_NUM) {
+ DBG("Unkown register: reg: %d", reg);
+ return -EINVAL;
+ }
+
+ uda1341_write_reg_cache(codec, reg, value);
+
+ switch (reg) {
+ case UDA1341_STATUS0:
+ case UDA1341_STATUS1:
+ addr = UDA1341_STATUS_ADDR;
+ break;
+ case UDA1341_DATA000:
+ case UDA1341_DATA001:
+ case UDA1341_DATA010:
+ addr = UDA1341_DATA0_ADDR;
+ break;
+ case UDA1341_DATA1:
+ addr = UDA1341_DATA1_ADDR;
+ break;
+ default:
+ /* It's an extended address register */
+ addr = (reg | UDA1341_EXTADDR_PREFIX);
+
+ ret = l3_write((struct uda1341_platform_data *)
+ codec->control_data,
+ UDA1341_DATA0_ADDR, &addr, 1);
+ if (ret != 1)
+ return -EIO;
+
+ addr = UDA1341_DATA0_ADDR;
+ data = (value | UDA1341_EXTDATA_PREFIX);
+ break;
+ }
+
+ ret = l3_write((struct uda1341_platform_data *) codec->control_data,
+ addr, &data, 1);
+ if (ret != 1)
+ return -EIO;
+
+ return 0;
+}
+
+static inline void uda1341_reset(struct snd_soc_codec *codec)
+{
+ u8 reset_reg = uda1341_read_reg_cache(codec, UDA1341_STATUS0);
+ uda1341_write(codec, UDA1341_STATUS0, reset_reg | (1<<6));
+ mdelay(1);
+ uda1341_write(codec, UDA1341_STATUS0, reset_reg & ~(1<<6));
+}
+
+static int uda1341_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u8 mute_reg = uda1341_read_reg_cache(codec, UDA1341_DATA010);
+
+ DBG("mute: %d", mute);
+
+ if (mute)
+ mute_reg |= (1<<2);
+ else
+ mute_reg &= ~(1<<2);
+
+ uda1341_write(codec, UDA1341_DATA010, mute_reg & ~(1<<2));
+
+ return 0;
+}
+
+static int uda1341_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 uda1341_priv *uda1341 = codec->private_data;
+
+ u8 hw_params = uda1341_read_reg_cache(codec, UDA1341_STATUS0);
+ hw_params &= STATUS0_SYSCLK_MASK;
+ hw_params &= STATUS0_DAIFMT_MASK;
+
+ DBG("sysclk: %d, rate:%d", uda1341->sysclk, params_rate(params));
+
+ /* set SYSCLK / fs ratio */
+ switch (uda1341->sysclk / params_rate(params)) {
+ case 512:
+ break;
+ case 384:
+ hw_params |= (1<<4);
+ break;
+ case 256:
+ hw_params |= (1<<5);
+ break;
+ default:
+ return -EINVAL;
+ break;
+ }
+
+ DBG("dai_fmt: %d, params_format:%d", uda1341->dai_fmt,
+ params_format(params));
+
+
+ /* set DAI format and word length */
+ switch (uda1341->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:
+ return -EINVAL;
+ }
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ hw_params |= (1<<3);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ uda1341_write(codec, UDA1341_STATUS0, hw_params);
+
+ return 0;
+}
+
+static int uda1341_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 uda1341_priv *uda1341 = codec->private_data;
+
+ DBG("clk_id: %d, freq: %d, dir: %d", clk_id, freq, dir);
+
+ /* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable
+ we'll error out on set_hw_params if it's not OK */
+ if ((freq >= (256 * 8000)) && (freq <= (512 * 48000))) {
+ uda1341->sysclk = freq;
+ return 0;
+ }
+
+ return -EINVAL;
+}
+
+static int uda1341_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct uda1341_priv *uda1341 = codec->private_data;
+
+ DBG("fmt: %08X", fmt);
+
+ /* codec supports only full slave mode */
+ if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS)
+ return -EINVAL;
+
+ /* no support for clock inversion */
+ if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF)
+ 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 */
+ uda1341->dai_fmt = fmt;
+
+ return 0;
+}
+
+static int uda1341_dapm_event(struct snd_soc_codec *codec, int event)
+{
+ u8 reg;
+
+ DBG("event: %08X", event);
+
+ switch (event) {
+ case SNDRV_CTL_POWER_D0: /* full On */
+ /* ADC, DAC on */
+ reg = uda1341_read_reg_cache(codec, UDA1341_STATUS1);
+ uda1341_write(codec, UDA1341_STATUS1, reg | 0x03);
+ break;
+ case SNDRV_CTL_POWER_D1: /* partial On */
+ case SNDRV_CTL_POWER_D2: /* partial On */
+ case SNDRV_CTL_POWER_D3hot: /* Off, with power */
+ break;
+ case SNDRV_CTL_POWER_D3cold: /* Off, without power */
+ /* mute, ADC, DAC power off */
+ reg = uda1341_read_reg_cache(codec, UDA1341_STATUS1);
+ uda1341_write(codec, UDA1341_STATUS1, reg & ~(0x03));
+ break;
+ }
+ return 0;
+}
+
+static const char *uda1341_dsp_setting[] = {"Flat", "Minimum1",
+ "Minimum2", "Maximum"};
+static const char *uda1341_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
+static const char *uda1341_mixmode[] = {"DD", "Input 2",
+ "Input 2", "Digital mixer"};
+
+static const struct soc_enum uda1341_mixer_enum[] = {
+SOC_ENUM_SINGLE(UDA1341_DATA010, 0, 0x03, uda1341_dsp_setting),
+SOC_ENUM_SINGLE(UDA1341_DATA010, 3, 0x03, uda1341_deemph),
+SOC_ENUM_SINGLE(UDA1341_EA010, 0, 0x03, uda1341_mixmode),
+};
+
+static const struct snd_kcontrol_new uda1341_snd_controls[] = {
+SOC_SINGLE("Playback Volume", UDA1341_DATA000, 0, 0x3F, 1),
+SOC_SINGLE("Mic gain", UDA1341_EA010, 2, 0x07, 0),
+SOC_SINGLE("Channel 1 mixer gain", UDA1341_EA000, 0, 0x1F, 1),
+SOC_SINGLE("Channgel 2 mixer gain", UDA1341_EA001, 0, 0x1F, 1),
+SOC_SINGLE("Input channel 2 amp gain", UDA1341_EA101, 0, 0x1F, 0),
+
+SOC_SINGLE("Bass boost", UDA1341_DATA001, 2, 0xF, 0),
+SOC_SINGLE("Treble", UDA1341_DATA001, 0, 3, 0),
+
+SOC_ENUM("DSP setting", uda1341_mixer_enum[0]),
+SOC_ENUM("Playback De-emphasis", uda1341_mixer_enum[1]),
+SOC_ENUM("Mixer mode", uda1341_mixer_enum[2]),
+
+/* This should be an ext control with own handler, if one wants
+ to set the values in 0.5dB steps instead of 3dB */
+SOC_SINGLE("AGC output level", UDA1341_EA110, 0, 0x03, 1),
+SOC_SINGLE("AGC time const", UDA1341_EA110, 2, 0x07, 0),
+
+SOC_SINGLE("DAC Gain switch", UDA1341_STATUS1, 6, 1, 0),
+SOC_SINGLE("ADC Gain switch", UDA1341_STATUS1, 5, 1, 0),
+SOC_SINGLE("ADC Polarity switch", UDA1341_STATUS1, 4, 1, 0),
+SOC_SINGLE("DAC Polarity switch", UDA1341_STATUS1, 3, 1, 0),
+SOC_SINGLE("Double speed playback switch", UDA1341_STATUS1, 2, 1, 0),
+};
+
+static int uda1341_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(uda1341_snd_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&uda1341_snd_controls[i],
+ codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+struct snd_soc_dai uda1341_dai = {
+ .name = "UDA1341",
+ /* playback capabilities */
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = UDA1341_RATES,
+ .formats = UDA1341_FORMATS,
+ },
+ /* capture capabilities */
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = UDA1341_RATES,
+ .formats = UDA1341_FORMATS,
+ },
+ /* pcm operations */
+ .ops = {
+ .hw_params = uda1341_hw_params,
+ },
+ /* DAI operations */
+ .dai_ops = {
+ .digital_mute = uda1341_mute,
+ .set_sysclk = uda1341_set_dai_sysclk,
+ .set_fmt = uda1341_set_dai_fmt,
+ }
+};
+EXPORT_SYMBOL(uda1341_dai);
+
+
+static int uda1341_soc_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ struct uda1341_priv *uda1341;
+ void *codec_setup_data = socdev->codec_data;
+ int ret = -ENOMEM;
+ struct uda1341_platform_data *pd;
+
+ printk(KERN_INFO "UDA1341 SoC Audio Codec\n");
+
+ socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (socdev->codec == NULL)
+ return ret;
+
+ codec = socdev->codec;
+
+ uda1341 = kzalloc(sizeof(struct uda1341_priv), GFP_KERNEL);
+ if (uda1341 == NULL)
+ goto priv_err;
+
+ codec->private_data = uda1341;
+
+ codec->reg_cache = kmemdup(uda1341_reg, sizeof(uda1341_reg),
+ GFP_KERNEL);
+
+ if (codec->reg_cache == NULL)
+ goto reg_err;
+
+ mutex_init(&codec->mutex);
+
+ codec->reg_cache_size = sizeof(uda1341_reg);
+ codec->reg_cache_step = 1;
+
+ codec->name = "UDA1341";
+ codec->owner = THIS_MODULE;
+ codec->dai = &uda1341_dai;
+ codec->num_dai = 1;
+ codec->read = uda1341_read_reg_cache;
+ codec->write = uda1341_write;
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ if (!codec_setup_data) {
+ printk(KERN_ERR "UDA1341 SoC codec: "
+ "missing L3 bitbang function\n");
+ ret = -ENODEV;
+ goto pcm_err;
+ }
+
+ codec->control_data = codec_setup_data;
+ pd = (struct uda1341_platform_data *) codec_setup_data;
+
+ if (pd->power)
+ pd->power(1);
+
+ uda1341_reset(codec);
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ printk(KERN_ERR "UDA1341: failed to register pcms\n");
+ goto pcm_err;
+ }
+
+ ret = uda1341_add_controls(codec);
+ if (ret < 0) {
+ printk(KERN_ERR "UDA1341: failed to register controls\n");
+ goto pcm_err;
+ }
+
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "UDA1341: 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 uda1341_soc_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ struct uda1341_platform_data *pd;
+
+ uda1341_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+
+ pd = (struct uda1341_platform_data *) codec->control_data;
+ if (pd->power)
+ pd->power(0);
+
+ 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 uda1341_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;
+ struct uda1341_platform_data *pd;
+
+ uda1341_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+
+ pd = (struct uda1341_platform_data *) codec->control_data;
+ if (pd->power)
+ pd->power(0);
+
+ return 0;
+}
+
+static int uda1341_soc_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ struct uda1341_platform_data *pd;
+ int i;
+ u8 *cache = codec->reg_cache;
+
+ pd = (struct uda1341_platform_data *) codec->control_data;
+ if (pd->power)
+ pd->power(1);
+
+ /* Sync reg_cache with the hardware */
+ for (i = 0; i < ARRAY_SIZE(uda1341_reg); i++)
+ codec->write(codec, i, *cache++);
+ uda1341_dapm_event(codec, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif /* CONFIG_PM */
+
+struct snd_soc_codec_device soc_codec_dev_uda1341 = {
+ .probe = uda1341_soc_probe,
+ .remove = uda1341_soc_remove,
+#if defined(CONFIG_PM)
+ .suspend = uda1341_soc_suspend,
+ .resume = uda1341_soc_resume,
+#endif /* CONFIG_PM */
+};
+EXPORT_SYMBOL(soc_codec_dev_uda1341);
+
+MODULE_DESCRIPTION("UDA1341 ALSA soc codec driver");
+MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell at evolware.org>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/uda1341/uda1341.h b/sound/soc/codecs/uda1341/uda1341.h
new file mode 100644
index 0000000..5eccfc4
--- /dev/null
+++ b/sound/soc/codecs/uda1341/uda1341.h
@@ -0,0 +1,47 @@
+/*
+ * uda1341.h -- UDA1341 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 _UDA1341_H
+#define _UDA1341_H
+
+#define UDA1341_L3ADDR 5
+#define UDA1341_DATA0_ADDR ((UDA1341_L3ADDR << 2) | 0)
+#define UDA1341_DATA1_ADDR ((UDA1341_L3ADDR << 2) | 1)
+#define UDA1341_STATUS_ADDR ((UDA1341_L3ADDR << 2) | 2)
+
+#define UDA1341_EXTADDR_PREFIX 0xC0
+#define UDA1341_EXTDATA_PREFIX 0xE0
+
+/* UDA1341 registers */
+#define UDA1341_EA000 0
+#define UDA1341_EA001 1
+#define UDA1341_EA010 2
+#define UDA1341_EA011 3
+#define UDA1341_EA100 4
+#define UDA1341_EA101 5
+#define UDA1341_EA110 6
+#define UDA1341_EA111 7
+#define UDA1341_STATUS0 8
+#define UDA1341_STATUS1 9
+#define UDA1341_DATA000 10
+#define UDA1341_DATA001 11
+#define UDA1341_DATA010 12
+#define UDA1341_DATA1 13
+
+#define UDA1341_REGS_NUM 14
+
+#define STATUS0_DAIFMT_MASK (~(7<<1))
+#define STATUS0_SYSCLK_MASK (~(3<<4))
+
+extern struct snd_soc_dai uda1341_dai;
+extern struct snd_soc_codec_device soc_codec_dev_uda1341;
+
+#endif /* _UDA1341_H */
diff --git a/sound/soc/codecs/uda1341/uda1341_platform_data.h b/sound/soc/codecs/uda1341/uda1341_platform_data.h
new file mode 100644
index 0000000..26081cb
--- /dev/null
+++ b/sound/soc/codecs/uda1341/uda1341_platform_data.h
@@ -0,0 +1,18 @@
+#ifndef _UDA1341_PLATFORM_DATA_H_
+#define _UDA1341_PLATFORM_DATA_H_ 1
+
+struct uda1341_platform_data {
+ void (*setdat) (struct uda1341_platform_data *, int);
+ void (*setclk) (struct uda1341_platform_data *, int);
+ void (*setmode) (struct uda1341_platform_data *, int);
+ int data_hold;
+ int data_setup;
+ int clock_high;
+ int mode_hold;
+ int mode;
+ int mode_setup;
+ void *priv;
+ void (*power) (int);
+};
+
+#endif
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
index b9f2353..cc8fa5e 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -16,7 +16,7 @@ config SND_S3C2443_SOC_AC97
tristate
select AC97_BUS
select SND_SOC_AC97_BUS
-
+
config SND_S3C24XX_SOC_NEO1973_WM8753
tristate "SoC I2S Audio support for NEO1973 - WM8753"
depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA01
@@ -44,3 +44,8 @@ config SND_S3C24XX_SOC_LN2440SBC_ALC650
Say Y if you want to add support for SoC audio on ln2440sbc
with the ALC650.
+config SND_S3C24XX_SOC_S3C24XX_UDA1341
+ tristate "SoC I2S Audio support UDA1341 wired to a S3C24XX"
+ depends on SND_S3C24XX_SOC
+ select SND_S3C24XX_SOC_I2S
+ select SND_SOC_UDA1341
\ No newline at end of file
diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile
index 0aa5fb0..155f5a4 100644
--- a/sound/soc/s3c24xx/Makefile
+++ b/sound/soc/s3c24xx/Makefile
@@ -13,7 +13,9 @@ obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o
snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o
snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o
snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o
+snd-soc-s3c24xx-uda1341-objs := s3c24xx_uda1341.o
obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o
obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o
+obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA1341) += snd-soc-s3c24xx-uda1341.o
diff --git a/sound/soc/s3c24xx/s3c24xx_uda1341.c b/sound/soc/s3c24xx/s3c24xx_uda1341.c
new file mode 100644
index 0000000..fc3c13e
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c24xx_uda1341.c
@@ -0,0 +1,314 @@
+/*
+ * Modifications by Christian Pellegrin <chripell at evolware.org>
+ *
+ * s3c24xx_uda1341.c -- S3C24XX_UDA1341 ALSA SoC Audio board 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.
+ */
+
+#include <linux/module.h>
+#include <linux/clk.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/s3c24xx_uda1341.h>
+
+#include <asm/mach-types.h>
+#include <asm/plat-s3c24xx/regs-iis.h>
+#include <mach/regs-gpio.h>
+#include <mach/regs-gpioj.h>
+#include <mach/hardware.h>
+
+#include "../codecs/uda1341/uda1341.h"
+#include "../codecs/uda1341/uda1341_platform_data.h"
+#include "s3c24xx-pcm.h"
+#include "s3c24xx-i2s.h"
+
+/*#define S3C24XX_UDA1341_DEBUG 1*/
+#ifdef S3C24XX_UDA1341_DEBUG
+#define DBG(x...) printk(KERN_DEBUG "s3c24xx-i2s: " x)
+#else
+#define DBG(x...)
+#endif
+
+static struct clk *xtal;
+static struct clk *pclk;
+
+static unsigned long s3c24xx_uda1341_calc_error(unsigned long rate,
+ unsigned long clk_rate,
+ unsigned int div,
+ unsigned int fs)
+{
+ long err;
+
+ err = clk_rate / (div * fs);
+ err -= rate;
+ if (err < 0)
+ err = -err;
+ DBG("rate %lu clk %lu div %u fs %u err %ld\n",
+ rate, clk_rate, div, fs, err);
+ return err;
+}
+
+static int s3c24xx_uda1341_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_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ unsigned int clk = 0;
+ unsigned int div = 0, cdiv;
+ int ret = 0;
+ int clk_source, fs_mode;
+ unsigned long mpllin_rate = clk_get_rate(xtal);
+ unsigned long pclk_rate = clk_get_rate(pclk);
+ unsigned long rate = params_rate(params);
+ unsigned long err, cerr;
+
+ DBG("mpllin %ld pclk %ld rate %lu\n", mpllin_rate, pclk_rate, rate);
+
+ div = pclk_rate / (256 * rate);
+ if (div == 0)
+ div = 1;
+ if (div > 32)
+ div = 32;
+ err = s3c24xx_uda1341_calc_error(rate, pclk_rate, div, 256);
+ fs_mode = S3C2410_IISMOD_256FS;
+ clk_source = S3C24XX_CLKSRC_PCLK;
+
+ if (div < 32) {
+ cdiv = div + 1;
+ cerr = s3c24xx_uda1341_calc_error(rate, pclk_rate, cdiv, 256);
+ if (cerr < err) {
+ err = cerr;
+ div = cdiv;
+ }
+ }
+
+ cdiv = pclk_rate / (384 * rate);
+ if (cdiv == 0)
+ cdiv = 1;
+ if (cdiv > 32)
+ cdiv = 32;
+ cerr = s3c24xx_uda1341_calc_error(rate, pclk_rate, cdiv, 384);
+ if (cerr < err) {
+ err = cerr;
+ div = cdiv;
+ fs_mode = S3C2410_IISMOD_384FS;
+ }
+
+ if (cdiv < 32) {
+ cdiv = cdiv + 1;
+ cerr = s3c24xx_uda1341_calc_error(rate, pclk_rate, cdiv, 384);
+ if (cerr < err) {
+ err = cerr;
+ div = cdiv;
+ fs_mode = S3C2410_IISMOD_384FS;
+ }
+ }
+
+ cerr = s3c24xx_uda1341_calc_error(rate, mpllin_rate, 1, 256);
+ if (cerr < err) {
+ err = cerr;
+ fs_mode = S3C2410_IISMOD_256FS;
+ clk_source = S3C24XX_CLKSRC_MPLL;
+ }
+
+ cerr = s3c24xx_uda1341_calc_error(rate, mpllin_rate, 1, 384);
+ if (cerr < err) {
+ err = cerr;
+ fs_mode = S3C2410_IISMOD_384FS;
+ clk_source = S3C24XX_CLKSRC_MPLL;
+ }
+
+ clk = (fs_mode == S3C2410_IISMOD_384FS ? 384 : 256) * rate;
+ div = div - 1;
+ DBG("Will use: %s %s %d sysclk %d err %ld\n",
+ fs_mode == S3C2410_IISMOD_384FS ? "384FS" : "256FS",
+ clk_source == S3C24XX_CLKSRC_MPLL ? "MPLLin" : "PCLK",
+ div, clk, err);
+
+ ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, clk_source , clk,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
+ fs_mode);
+ if (ret < 0)
+ return ret;
+
+ ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK,
+ S3C2410_IISMOD_32FS);
+ if (ret < 0)
+ return ret;
+
+ ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
+ S3C24XX_PRESCALE(div, div));
+ if (ret < 0)
+ return ret;
+
+ /* set the codec system clock for DAC and ADC */
+ ret = codec_dai->dai_ops.set_sysclk(codec_dai, 0, clk,
+ SND_SOC_CLOCK_OUT);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops s3c24xx_uda1341_ops = {
+ .hw_params = s3c24xx_uda1341_hw_params,
+};
+
+static struct snd_soc_dai_link s3c24xx_uda1341_dai_link = {
+ .name = "UDA1341",
+ .stream_name = "UDA1341",
+ .codec_dai = &uda1341_dai,
+ .cpu_dai = &s3c24xx_i2s_dai,
+ .ops = &s3c24xx_uda1341_ops,
+};
+
+static struct snd_soc_machine snd_soc_machine_s3c24xx_uda1341 = {
+ .name = "S3C24XX_UDA1341",
+ .dai_link = &s3c24xx_uda1341_dai_link,
+ .num_links = 1,
+};
+
+static struct s3c24xx_uda1341_platform_data *s3c24xx_uda1341_l3_pins;
+
+static void setdat(struct uda1341_platform_data *p, int v)
+{
+ s3c2410_gpio_setpin(s3c24xx_uda1341_l3_pins->l3_data, v > 0);
+ s3c2410_gpio_cfgpin(s3c24xx_uda1341_l3_pins->l3_data,
+ S3C2410_GPIO_OUTPUT);
+}
+
+static void setclk(struct uda1341_platform_data *p, int v)
+{
+ s3c2410_gpio_setpin(s3c24xx_uda1341_l3_pins->l3_clk, v > 0);
+ s3c2410_gpio_cfgpin(s3c24xx_uda1341_l3_pins->l3_clk,
+ S3C2410_GPIO_OUTPUT);
+}
+
+static void setmode(struct uda1341_platform_data *p, int v)
+{
+ s3c2410_gpio_setpin(s3c24xx_uda1341_l3_pins->l3_mode, v > 0);
+ s3c2410_gpio_cfgpin(s3c24xx_uda1341_l3_pins->l3_mode,
+ S3C2410_GPIO_OUTPUT);
+}
+
+static void s3c24xx_uda1341_power(int en)
+{
+ if (s3c24xx_uda1341_l3_pins->power)
+ s3c24xx_uda1341_l3_pins->power(en);
+}
+
+static struct uda1341_platform_data s3c24xx_uda1341 = {
+ .setdat = setdat,
+ .setclk = setclk,
+ .setmode = setmode,
+ .data_hold = 1,
+ .data_setup = 1,
+ .clock_high = 1,
+ .mode_hold = 1,
+ .mode = 1,
+ .mode_setup = 1,
+ .power = s3c24xx_uda1341_power,
+};
+
+static struct snd_soc_device s3c24xx_uda1341_snd_devdata = {
+ .machine = &snd_soc_machine_s3c24xx_uda1341,
+ .platform = &s3c24xx_soc_platform,
+ .codec_dev = &soc_codec_dev_uda1341,
+ .codec_data = &s3c24xx_uda1341,
+};
+
+static struct platform_device *s3c24xx_uda1341_snd_device;
+
+static int s3c24xx_uda1341_probe(struct platform_device *pdev)
+{
+ int ret;
+
+ printk(KERN_INFO "S3C24XX_UDA1341 SoC Audio driver\n");
+
+ s3c24xx_uda1341_l3_pins = pdev->dev.platform_data;
+ if (s3c24xx_uda1341_l3_pins == NULL) {
+ printk(KERN_ERR "S3C24XX_UDA1341 SoC Audio: "
+ "unable to find platform data\n");
+ return -ENODEV;
+ }
+
+ s3c24xx_uda1341_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!s3c24xx_uda1341_snd_device) {
+ printk(KERN_ERR "S3C24XX_UDA1341 SoC Audio: "
+ "Unable to register\n");
+ return -ENOMEM;
+ }
+
+ platform_set_drvdata(s3c24xx_uda1341_snd_device,
+ &s3c24xx_uda1341_snd_devdata);
+ s3c24xx_uda1341_snd_devdata.dev = &s3c24xx_uda1341_snd_device->dev;
+ ret = platform_device_add(s3c24xx_uda1341_snd_device);
+
+ if (ret) {
+ printk(KERN_ERR "S3C24XX_UDA1341 SoC Audio: Unable to add\n");
+ platform_device_put(s3c24xx_uda1341_snd_device);
+ }
+
+ xtal = clk_get(NULL, "xtal");
+ pclk = clk_get(NULL, "pclk");
+
+ return ret;
+}
+
+static int s3c24xx_uda1341_remove(struct platform_device *pdev)
+{
+ platform_device_unregister(s3c24xx_uda1341_snd_device);
+ clk_put(xtal);
+ clk_put(pclk);
+ return 0;
+}
+
+static struct platform_driver s3c24xx_uda1341_driver = {
+ .probe = s3c24xx_uda1341_probe,
+ .remove = s3c24xx_uda1341_remove,
+ .driver = {
+ .name = "s3c24xx_uda1341",
+ .owner = THIS_MODULE,
+ },
+};
+
+static int __init s3c24xx_uda1341_init(void)
+{
+ return platform_driver_register(&s3c24xx_uda1341_driver);
+}
+
+static void __exit s3c24xx_uda1341_exit(void)
+{
+ platform_driver_unregister(&s3c24xx_uda1341_driver);
+}
+
+
+module_init(s3c24xx_uda1341_init);
+module_exit(s3c24xx_uda1341_exit);
+
+MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell at evolware.org>");
+MODULE_DESCRIPTION("S3C24XX_UDA1341 ALSA SoC audio driver");
+MODULE_LICENSE("GPL");
--
1.4.4.4
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