Sound card definition for for s3c24xx with uda134x
Signed-off-by: Christian Pellegrin <chripell(a)fsfe.org>
---
include/sound/s3c24xx_uda134x.h | 14 ++
sound/soc/s3c24xx/Kconfig | 5 +
sound/soc/s3c24xx/Makefile | 2 +
sound/soc/s3c24xx/s3c24xx_uda134x.c | 374 +++++++++++++++++++++++++++++++++++
4 files changed, 395 insertions(+), 0 deletions(-)
--
Generated on 20081115 against f7e1798a78a30bae1cf3ebc3e1b6f7c1bac57756
--
diff --git a/include/sound/s3c24xx_uda134x.h b/include/sound/s3c24xx_uda134x.h
new file mode 100644
index 0000000..33df4cb
--- /dev/null
+++ b/include/sound/s3c24xx_uda134x.h
@@ -0,0 +1,14 @@
+#ifndef _S3C24XX_UDA134X_H_
+#define _S3C24XX_UDA134X_H_ 1
+
+#include <sound/uda134x.h>
+
+struct s3c24xx_uda134x_platform_data {
+ int l3_clk;
+ int l3_mode;
+ int l3_data;
+ void (*power) (int);
+ int model;
+};
+
+#endif
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
index b9f2353..fcd03ac 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -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_UDA134X
+ tristate "SoC I2S Audio support UDA134X wired to a S3C24XX"
+ depends on SND_S3C24XX_SOC
+ select SND_S3C24XX_SOC_I2S
+ select SND_SOC_UDA134X
diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile
index 0aa5fb0..96b3f3f 100644
--- a/sound/soc/s3c24xx/Makefile
+++ b/sound/soc/s3c24xx/Makefile
@@ -13,7 +13,9 @@ obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o
snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o
snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o
snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o
+snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o
obj-$(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_UDA134X) += snd-soc-s3c24xx-uda134x.o
diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c
new file mode 100644
index 0000000..29a6813
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c24xx_uda134x.c
@@ -0,0 +1,374 @@
+/*
+ * Modifications by Christian Pellegrin <chripell(a)evolware.org>
+ *
+ * s3c24xx_uda134x.c -- S3C24XX_UDA134X 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 <linux/mutex.h>
+#include <linux/gpio.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/s3c24xx_uda134x.h>
+#include <sound/uda134x.h>
+
+#include <asm/plat-s3c24xx/regs-iis.h>
+
+#include "s3c24xx-pcm.h"
+#include "s3c24xx-i2s.h"
+#include "../codecs/uda134x_codec.h"
+
+
+/* #define ENFORCE_RATES 1 */
+/*
+ Unfortunately the S3C24XX in master mode has a limited capacity of
+ generating the clock for the codec. If you define this only rates
+ that are really available will be enforced. But be careful, most
+ user level application just want the usual sampling frequencies (8,
+ 11.025, 22.050, 44.1 kHz) and anyway resampling is a costly
+ operation for embedded systems. So if you aren't very lucky or your
+ hardware engineer wasn't very forward-looking it's better to leave
+ this undefined. If you do so an approximate value for the requested
+ sampling rate in the range -/+ 5% will be chosen. If this in not
+ possible an error will be returned.
+*/
+
+static struct clk *xtal;
+static struct clk *pclk;
+/* this is need because we don't have a place where to keep the
+ * pointers to the clocks in each substream. We get the clocks only
+ * when we are actually using them so we don't block stuff like
+ * frequency change or oscillator power-off */
+static int clk_users;
+static DEFINE_MUTEX(clk_lock);
+
+static unsigned int rates[33 * 2];
+#ifdef ENFORCE_RATES
+static struct snd_pcm_hw_constraint_list hw_constraints_rates = {
+ .count = ARRAY_SIZE(rates),
+ .list = rates,
+ .mask = 0,
+};
+#endif
+
+static struct platform_device *s3c24xx_uda134x_snd_device;
+
+int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+#ifdef ENFORCE_RATES
+ struct snd_pcm_runtime *runtime = substream->runtime;;
+#endif
+
+ mutex_lock(&clk_lock);
+ pr_debug("%s %d\n", __func__, clk_users);
+ if (clk_users == 0) {
+ xtal = clk_get(&s3c24xx_uda134x_snd_device->dev, "xtal");
+ if (!xtal) {
+ printk(KERN_ERR "%s cannot get xtal\n", __func__);
+ ret = -EBUSY;
+ } else {
+ pclk = clk_get(&s3c24xx_uda134x_snd_device->dev,
+ "pclk");
+ if (!pclk) {
+ printk(KERN_ERR "%s cannot get pclk\n",
+ __func__);
+ clk_put(xtal);
+ ret = -EBUSY;
+ }
+ }
+ if (!ret) {
+ int i, j;
+
+ for (i = 0; i < 2; i++) {
+ int fs = i ? 256 : 384;
+
+ rates[i*33] = clk_get_rate(xtal) / fs;
+ for (j = 1; j < 33; j++)
+ rates[i*33 + j] = clk_get_rate(pclk) /
+ (j * fs);
+ }
+ }
+ }
+ clk_users += 1;
+ mutex_unlock(&clk_lock);
+ if (!ret) {
+#ifdef ENFORCE_RATES
+ ret = snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &hw_constraints_rates);
+ if (ret < 0)
+ printk(KERN_ERR "%s cannot set constraints\n",
+ __func__);
+#endif
+ }
+ return ret;
+}
+
+void s3c24xx_uda134x_shutdown(struct snd_pcm_substream *substream)
+{
+ mutex_lock(&clk_lock);
+ pr_debug("%s %d\n", __func__, clk_users);
+ clk_users -= 1;
+ if (clk_users == 0) {
+ clk_put(xtal);
+ xtal = NULL;
+ clk_put(pclk);
+ pclk = NULL;
+ }
+ mutex_unlock(&clk_lock);
+}
+
+static int s3c24xx_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_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ unsigned int clk = 0;
+ int ret = 0;
+ int clk_source, fs_mode;
+ unsigned long rate = params_rate(params);
+ long err, cerr;
+ unsigned int div;
+ int i, bi;
+
+ err = 999999;
+ bi = 0;
+ for (i = 0; i < 2*33; i++) {
+ cerr = rates[i] - rate;
+ if (cerr < 0)
+ cerr = -cerr;
+ if (cerr < err) {
+ err = cerr;
+ bi = i;
+ }
+ }
+ if (bi / 33 == 1)
+ fs_mode = S3C2410_IISMOD_256FS;
+ else
+ fs_mode = S3C2410_IISMOD_384FS;
+ if (bi % 33 == 0) {
+ clk_source = S3C24XX_CLKSRC_MPLL;
+ div = 1;
+ } else {
+ clk_source = S3C24XX_CLKSRC_PCLK;
+ div = bi % 33;
+ }
+ pr_debug("%s desired rate %lu, %d\n", __func__, rate, bi);
+
+ clk = (fs_mode == S3C2410_IISMOD_384FS ? 384 : 256) * rate;
+ pr_debug("%s will use: %s %s %d sysclk %d err %ld\n", __func__,
+ fs_mode == S3C2410_IISMOD_384FS ? "384FS" : "256FS",
+ clk_source == S3C24XX_CLKSRC_MPLL ? "MPLLin" : "PCLK",
+ div, clk, err);
+
+ if ((err * 100 / rate) > 5) {
+ printk(KERN_ERR "S3C24XX_UDA134X: effective frequency "
+ "too different from desired (%ld%%)\n",
+ err * 100 / rate);
+ return -EINVAL;
+ }
+
+ 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_uda134x_ops = {
+ .startup = s3c24xx_uda134x_startup,
+ .shutdown = s3c24xx_uda134x_shutdown,
+ .hw_params = s3c24xx_uda134x_hw_params,
+};
+
+static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = {
+ .name = "UDA134X",
+ .stream_name = "UDA134X",
+ .codec_dai = &uda134x_dai,
+ .cpu_dai = &s3c24xx_i2s_dai,
+ .ops = &s3c24xx_uda134x_ops,
+};
+
+static struct snd_soc_machine snd_soc_machine_s3c24xx_uda134x = {
+ .name = "S3C24XX_UDA134X",
+ .dai_link = &s3c24xx_uda134x_dai_link,
+ .num_links = 1,
+};
+
+static struct s3c24xx_uda134x_platform_data *s3c24xx_uda134x_l3_pins;
+
+static void setdat(int v)
+{
+ gpio_set_value(s3c24xx_uda134x_l3_pins->l3_data, v > 0);
+}
+
+static void setclk(int v)
+{
+ gpio_set_value(s3c24xx_uda134x_l3_pins->l3_clk, v > 0);
+}
+
+static void setmode(int v)
+{
+ gpio_set_value(s3c24xx_uda134x_l3_pins->l3_mode, v > 0);
+}
+
+static struct uda134x_platform_data s3c24xx_uda134x = {
+ .l3 = {
+ .setdat = setdat,
+ .setclk = setclk,
+ .setmode = setmode,
+ .data_hold = 1,
+ .data_setup = 1,
+ .clock_high = 1,
+ .mode_hold = 1,
+ .mode = 1,
+ .mode_setup = 1,
+ },
+};
+
+static struct snd_soc_device s3c24xx_uda134x_snd_devdata = {
+ .machine = &snd_soc_machine_s3c24xx_uda134x,
+ .platform = &s3c24xx_soc_platform,
+ .codec_dev = &soc_codec_dev_uda134x,
+ .codec_data = &s3c24xx_uda134x,
+};
+
+static int s3c24xx_uda134x_setup_pin(int pin, char *fun)
+{
+ if (gpio_request(pin, "s3c24xx_uda134x") < 0) {
+ printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: "
+ "l3 %s pin already in use", fun);
+ return -EBUSY;
+ }
+ gpio_direction_output(pin, 0);
+ return 0;
+}
+
+static int s3c24xx_uda134x_probe(struct platform_device *pdev)
+{
+ int ret;
+
+ printk(KERN_INFO "S3C24XX_UDA134X SoC Audio driver\n");
+
+ s3c24xx_uda134x_l3_pins = pdev->dev.platform_data;
+ if (s3c24xx_uda134x_l3_pins == NULL) {
+ printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: "
+ "unable to find platform data\n");
+ return -ENODEV;
+ }
+ s3c24xx_uda134x.power = s3c24xx_uda134x_l3_pins->power;
+ s3c24xx_uda134x.model = s3c24xx_uda134x_l3_pins->model;
+
+ if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_data,
+ "data") < 0)
+ return -EBUSY;
+ if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_clk,
+ "clk") < 0) {
+ gpio_free(s3c24xx_uda134x_l3_pins->l3_data);
+ return -EBUSY;
+ }
+ if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_mode,
+ "mode") < 0) {
+ gpio_free(s3c24xx_uda134x_l3_pins->l3_data);
+ gpio_free(s3c24xx_uda134x_l3_pins->l3_clk);
+ return -EBUSY;
+ }
+
+ s3c24xx_uda134x_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!s3c24xx_uda134x_snd_device) {
+ printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: "
+ "Unable to register\n");
+ return -ENOMEM;
+ }
+
+ platform_set_drvdata(s3c24xx_uda134x_snd_device,
+ &s3c24xx_uda134x_snd_devdata);
+ s3c24xx_uda134x_snd_devdata.dev = &s3c24xx_uda134x_snd_device->dev;
+ ret = platform_device_add(s3c24xx_uda134x_snd_device);
+ if (ret) {
+ printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: Unable to add\n");
+ platform_device_put(s3c24xx_uda134x_snd_device);
+ }
+
+ return ret;
+}
+
+static int s3c24xx_uda134x_remove(struct platform_device *pdev)
+{
+ platform_device_unregister(s3c24xx_uda134x_snd_device);
+ gpio_free(s3c24xx_uda134x_l3_pins->l3_data);
+ gpio_free(s3c24xx_uda134x_l3_pins->l3_clk);
+ gpio_free(s3c24xx_uda134x_l3_pins->l3_mode);
+ return 0;
+}
+
+static struct platform_driver s3c24xx_uda134x_driver = {
+ .probe = s3c24xx_uda134x_probe,
+ .remove = s3c24xx_uda134x_remove,
+ .driver = {
+ .name = "s3c24xx_uda134x",
+ .owner = THIS_MODULE,
+ },
+};
+
+static int __init s3c24xx_uda134x_init(void)
+{
+ return platform_driver_register(&s3c24xx_uda134x_driver);
+}
+
+static void __exit s3c24xx_uda134x_exit(void)
+{
+ platform_driver_unregister(&s3c24xx_uda134x_driver);
+}
+
+
+module_init(s3c24xx_uda134x_init);
+module_exit(s3c24xx_uda134x_exit);
+
+MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell(a)evolware.org>");
+MODULE_DESCRIPTION("S3C24XX_UDA134X ALSA SoC audio driver");
+MODULE_LICENSE("GPL");
--
1.4.4.4
