ALSA SOC driver for s3c24xx with uda1341

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:

Generated on 20081108 against v2.6.27

Quoted 2 lines · Christian Pellegrin

This patch adds support for the UDA1341 codec and a sound card definition for one of these UDAs connected to the s3c24xx. It is

First up, thanks for doing this work - it'd be good to see this merged into ASoC.

Quoted 1 line

*heavily* based on the one made by Zoltan Devai but:

Ideally this should be submitted as multiple patches - at least one for the codec and one for the board, probably also one for the l3 interface.

Quoted 1 line

Generated on 20081108 against v2.6.27

Please generate patches against current git - the topic/asoc branch of:

	git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git

In general, once the merge window is past it's best to submit patches against at least -rc1, submitting against the previous release makes it much more likely that your patch will be out of date as has happened here.

sound/soc/codecs/Kconfig config SND_SOC_CS4270_VD33_ERRATA · line 55

Quoted 8 lines

+++ b/sound/soc/codecs/Kconfig

sound/soc/codecs/Kconfig +3 -0

@@ -50,3 +50,6 @@ config SND_SOC_CS4270_VD33_ERRATA
 config SND_SOC_TLV320AIC3X
 	tristate
 	depends on I2C
+
+config SND_SOC_UDA1341
+	tristate

You should also add this to SND_SOC_ALL_CODECS. There's a bunch of other things like this that have changed between 2.6.27 and the current code - for example, the bias level configuration.

sound/soc/codecs/Makefile obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o · line 26

Quoted 9 lines

sound/soc/codecs/Makefile +1 -0

index d7b97ab..cbace60 100644
@@ -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/

There doesn't seem to be much benefit to adding a subdirectory for two source files here.

sound/soc/codecs/uda1341/l3.c line 36

Quoted 10 lines

sound/soc/codecs/uda1341/l3.c new file +7 -0

 
+/*#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

Please use the standard pr_debug() macro rather than defining custom ones.

sound/soc/codecs/uda1341/l3.c line 40

Quoted 3 lines

sound/soc/codecs/uda1341/l3.c new file +3 -0

+#define setdat(adap, val)	(adap->setdat(adap, val))
+#define setclk(adap, val)	(adap->setclk(adap, val))
+#define setmode(adap, val)	(adap->setmode(adap, val))

These should be inline functions for type safety.

sound/soc/codecs/uda1341/l3.c

Quoted 7 lines

sound/soc/codecs/uda1341/l3.c new file +7 -0

+/*#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

Please use the standard pr_debug() macro.

sound/soc/codecs/uda1341/l3.c

Quoted 13 lines

sound/soc/codecs/uda1341/l3.c new file +13 -0

+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;
+	}

That should probably print the error unconditionally.

sound/soc/codecs/uda1341/l3.c

Quoted 11 lines

sound/soc/codecs/uda1341/l3.c new file +9 -0

+static int uda1341_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
 
+	/* set SYSCLK / fs ratio */
+	switch (uda1341->sysclk / params_rate(params)) {
 
+	default:
+		return -EINVAL;
+		break;
+	}

No need for the break here. It's probably best to log an explicit error saying why the failure occurred - otherwise it'll be a bit obscure to users what's going on. A similar comment applies to the other error cases.

sound/soc/codecs/uda1341/l3.c

Quoted 6 lines

sound/soc/codecs/uda1341/l3.c new file +5 -0

+static int uda1341_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+				  int clk_id, unsigned int freq, int dir)
+{
 
+	/* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable
+	we'll error out on set_hw_params if it's not OK */

This implies rather more flexibility than actually exists - hw_params() requires an exact multiplier here. You should probably add a note explaining that it's up to the machine driver to make sure that the rate is OK.

sound/soc/codecs/uda1341/l3.c

Quoted 2 lines

sound/soc/codecs/uda1341/l3.c new file +2 -0

+SOC_SINGLE("Channel 1 mixer gain", UDA1341_EA000, 0, 0x1F, 1),
+SOC_SINGLE("Channgel 2 mixer gain", UDA1341_EA001, 0, 0x1F, 1),

There's a typo here. Also, this and many of the other control names don't fit in with the ALSA control name spec so won't be displayed correctly by applications. For example, these should be "Volume" rather than "gain", and "switch" should be "Switch". There's documentation of the standard
names in:

Documentation/sound/alsa/ControlNames.txt

sound/soc/codecs/uda1341/l3.c

Quoted 2 lines

sound/soc/codecs/uda1341/l3.c new file +2 -0

+SOC_SINGLE("DAC Gain switch", UDA1341_STATUS1, 6, 1, 0),
+SOC_SINGLE("ADC Gain switch", UDA1341_STATUS1, 5, 1, 0),

What exactly do these controls do? "Gain switch" sounds wrong - I'd not expect the gain to be a boolean.

sound/soc/codecs/uda1341/l3.c

Quoted 12 lines

sound/soc/codecs/uda1341/l3.c new file +12 -0

+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);

I'd expect that dapm_event() (which is now called set_bias_level()) would be handling the power callback too.

sound/soc/codecs/uda1341/l3.c

Quoted 7 lines

sound/soc/codecs/uda1341/l3.c new file +7 -0

+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 */

The standard idiom for this is to have an else defining the functions to NULL pointers above and then no ifdef here.

sound/soc/codecs/uda1341/l3.c

Quoted 1 line

sound/soc/codecs/uda1341/l3.c new file +1 -0

+	void (*power) (int);

I'd really expect to see this being passed an argument specifying what it's interacting with.

sound/soc/codecs/uda1341/l3.c config SND_S3C2443_SOC_AC97 · line 22

Quoted 9 lines

sound/soc/codecs/uda1341/l3.c new file +1 -1

@@ -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

Random whitespace change here...

sound/soc/s3c24xx/s3c24xx_uda1341.c line 38

Quoted 9 lines

sound/soc/s3c24xx/s3c24xx_uda1341.c new file +6 -0

 
+/*#define S3C24XX_UDA1341_DEBUG 1*/
+#ifdef S3C24XX_UDA1341_DEBUG
+#define DBG(x...) printk(KERN_DEBUG "s3c24xx-i2s: " x)
+#else
+#define DBG(x...)
+#endif

pr_debug().

sound/soc/s3c24xx/s3c24xx_uda1341.c line 61

Quoted 3 lines

sound/soc/s3c24xx/s3c24xx_uda1341.c new file +3 -0

+static int s3c24xx_uda1341_hw_params(struct snd_pcm_substream *substream,
+                                       struct snd_pcm_hw_params *params)
+{

I can follow this function but it'd be nice to see more comments in here. It looks like you could clarify it by iterating over a table of possible clock sources rather than writing each out in code.

It also looks like this should be imposing constraints which prevent the configuration of different rates for the DAC and ADC - several existing codec drivers like the wm8903 provide examples of doing this.

sound/soc/s3c24xx/s3c24xx_uda1341.c line 150

Quoted 4 lines

sound/soc/s3c24xx/s3c24xx_uda1341.c new file +4 -0

+	ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, clk_source , clk,
+						SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;

You want to run this through scripts/checkpatch.pl.

sound/soc/s3c24xx/s3c24xx_uda1341.c line 201

Quoted 6 lines

sound/soc/s3c24xx/s3c24xx_uda1341.c new file +6 -0

+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);
+}

Is there any reason for setting the pins to output mode each time? It looks like you could just configure the mode once at startup and then only set the pin status here.

sound/soc/s3c24xx/s3c24xx_uda1341.c line 221

Quoted 5 lines

sound/soc/s3c24xx/s3c24xx_uda1341.c new file +5 -0

+static void s3c24xx_uda1341_power(int en)
+{
+	if (s3c24xx_uda1341_l3_pins->power)
+		s3c24xx_uda1341_l3_pins->power(en);
+}

This feels like it ought to be initialised conditionally rather than having the check for the pin it.

sound/soc/s3c24xx/s3c24xx_uda1341.c line 276

Quoted 4 lines

sound/soc/s3c24xx/s3c24xx_uda1341.c new file +3 -0

+	ret = platform_device_add(s3c24xx_uda1341_snd_device);
 
+	xtal = clk_get(NULL, "xtal");
+	pclk = clk_get(NULL, "pclk");

This should be done in the init function for the device and should really check the return value in case it can't get the clock for some reason. Ideally there'd be a dev there, but I'd need to check if the s3c24xx stuff does that.

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Ideally this should be submitted as multiple patches - at least one for the codec and one for the board, probably also one for the l3 interface.

Please generate patches against current git - the topic/asoc branch of:

You should also add this to SND_SOC_ALL_CODECS. There's a bunch of other things like this that have changed between 2.6.27 and the current code - for example, the bias level configuration.

There doesn't seem to be much benefit to adding a subdirectory for two source files here.

Please use the standard pr_debug() macro rather than defining custom ones.

These should be inline functions for type safety.

That should probably print the error unconditionally.

sound/soc/codecs/uda1341/uda1341.c line 186

Quoted 6 lines

sound/soc/codecs/uda1341/uda1341.c new file +5 -0

+     switch (uda1341->sysclk / params_rate(params)) {
 
+     default:
+             return -EINVAL;
+             break;
+     }

No need for the break here. It's probably best to log an explicit error saying why the failure occurred - otherwise it'll be a bit obscure to users what's going on. A similar comment applies to the other error cases.

Quoted 3 lines

sound/soc/codecs/uda1341/uda1341.c new file +3 -0

+static int uda1341_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+                               int clk_id, unsigned int freq, int dir)
+{

sound/soc/codecs/uda1341/uda1341.c line 232

Quoted 2 lines

sound/soc/codecs/uda1341/uda1341.c new file +2 -0

+     /* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable
+     we'll error out on set_hw_params if it's not OK */

This implies rather more flexibility than actually exists - hw_params() requires an exact multiplier here. You should probably add a note explaining that it's up to the machine driver to make sure that the rate is OK.

There's a typo here. Also, this and many of the other control names don't fit in with the ALSA control name spec so won't be displayed correctly by applications. For example, these should be "Volume" rather than "gain", and "switch" should be "Switch". There's documentation of the standard
names in:

Documentation/sound/alsa/ControlNames.txt

sound/soc/codecs/uda1341/uda1341.c line 321

Quoted 2 lines

sound/soc/codecs/uda1341/uda1341.c new file +2 -0

+SOC_SINGLE("DAC Gain switch", UDA1341_STATUS1, 6, 1, 0),
+SOC_SINGLE("ADC Gain switch", UDA1341_STATUS1, 5, 1, 0),

What exactly do these controls do? "Gain switch" sounds wrong - I'd not expect the gain to be a boolean.

sound/soc/codecs/uda1341/uda1341.c line 476

Quoted 3 lines

sound/soc/codecs/uda1341/uda1341.c new file +3 -0

+     pd = (struct uda1341_platform_data *) codec->control_data;
+     if (pd->power)
+             pd->power(0);

I'd expect that dapm_event() (which is now called set_bias_level()) would be handling the power callback too.

sound/soc/codecs/uda1341/uda1341.c line 531

Quoted 7 lines

sound/soc/codecs/uda1341/uda1341.c new file +7 -0

+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 */

The standard idiom for this is to have an else defining the functions to NULL pointers above and then no ifdef here.

Quoted 1 line

+ void (*power) (int);

I'd really expect to see this being passed an argument specifying what it's interacting with.

sound/soc/s3c24xx/Kconfig config SND_S3C2443_SOC_AC97 · line 22

Quoted 5 lines

sound/soc/s3c24xx/Kconfig +1 -1

-
+
 config SND_S3C24XX_SOC_NEO1973_WM8753
      tristate "SoC I2S Audio support for NEO1973 - WM8753"
      depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA01

Random whitespace change here...

sound/soc/s3c24xx/s3c24xx_uda1341.c line 61

Quoted 4 lines

sound/soc/s3c24xx/s3c24xx_uda1341.c new file +3 -0

+static int s3c24xx_uda1341_hw_params(struct snd_pcm_substream *substream,
+                                       struct snd_pcm_hw_params *params)
+{

I can follow this function but it'd be nice to see more comments in here. It looks like you could clarify it by iterating over a table of possible clock sources rather than writing each out in code.

It also looks like this should be imposing constraints which prevent the configuration of different rates for the DAC and ADC - several existing codec drivers like the wm8903 provide examples of doing this.

sound/soc/s3c24xx/s3c24xx_uda1341.c line 150

Quoted 4 lines

sound/soc/s3c24xx/s3c24xx_uda1341.c new file +4 -0

+     ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, clk_source , clk,
+                                             SND_SOC_CLOCK_IN);
+     if (ret < 0)
+             return ret;

You want to run this through scripts/checkpatch.pl.

sound/soc/s3c24xx/s3c24xx_uda1341.c line 201

Quoted 6 lines

sound/soc/s3c24xx/s3c24xx_uda1341.c new file +6 -0

+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);
+}

Is there any reason for setting the pins to output mode each time? It looks like you could just configure the mode once at startup and then only set the pin status here.

sound/soc/s3c24xx/s3c24xx_uda1341.c line 221

Quoted 6 lines

sound/soc/s3c24xx/s3c24xx_uda1341.c new file +5 -0

+static void s3c24xx_uda1341_power(int en)
+{
+     if (s3c24xx_uda1341_l3_pins->power)
+             s3c24xx_uda1341_l3_pins->power(en);
+}

This feels like it ought to be initialised conditionally rather than having the check for the pin it.

sound/soc/s3c24xx/s3c24xx_uda1341.c line 276

Quoted 4 lines

sound/soc/s3c24xx/s3c24xx_uda1341.c new file +3 -0

+     ret = platform_device_add(s3c24xx_uda1341_snd_device);
 
+     xtal = clk_get(NULL, "xtal");
+     pclk = clk_get(NULL, "pclk");

This should be done in the init function for the device and should really check the return value in case it can't get the clock for some reason. Ideally there'd be a dev there, but I'd need to check if the s3c24xx stuff does that.

On Mon, 10 Nov 2008 13:34:52 +0000

I'm equally interested in this patch, seeing as I
got an hp jornada 7xx with uda1344 waiting.

Have you any plans on creating an 134x seeing as there is very little difference between chipsets?

Quoted 8 lines · Kristoffer Ericson

On Mon, 10 Nov 2008 13:34:52 +0000

I'm equally interested in this patch, seeing as I
got an hp jornada 7xx with uda1344 waiting.

Have you any plans on creating an 134x seeing as there is very little difference between chipsets?

Hi,

I had a quick look at the UDA1344 data sheet. When operated in L3-mode it's basically a subset of the UDA1341 so I can do the changes. Anyway I'm not able to test them so could you give me a hand testing? Anyway if you won't use special mixer settings the patch I sent should work since the important configuration registers and the volume are the same for both chips.

Signature · 6 lines

-- 
Christian Pellegrin, see http://www.evolware.org/chri/
"Real Programmers don't play tennis, or any other sport which requires
you to change clothes. Mountain climbing is OK, and Real Programmers
wear their climbing boots to work in case a mountain should suddenly
spring up in the middle of the computer room."

sound/soc/s3c24xx/s3c24xx_uda1341.c line 276

Quoted 2 lines

sound/soc/s3c24xx/s3c24xx_uda1341.c new file +2 -0

+	xtal = clk_get(NULL, "xtal");
+	pclk = clk_get(NULL, "pclk");

This should be done in the init function for the device and should really check the return value in case it can't get the clock for some reason. Ideally there'd be a dev there, but I'd need to check if the s3c24xx stuff does that.

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 <[email protected]>

---
 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(-)

include/sound/s3c24xx_uda1341.h new file +22 -0

new file mode 100644
index 0000000..bd80bd3
@@ -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,
+	}
+};

include/sound/s3c24xx_uda1341.h new file +7 -0

+struct s3c24xx_uda1341_platform_data {
+	int l3_clk;
+	int l3_mode;
+	int l3_data;
+	void (*power) (int);
+};
 
+#endif

sound/soc/codecs/Kconfig +3 -0

index 1db04a2..4b483b7 100644
@@ -50,3 +50,6 @@ config SND_SOC_CS4270_VD33_ERRATA
 config SND_SOC_TLV320AIC3X
 	tristate
 	depends on I2C
+
+config SND_SOC_UDA1341
+	tristate

sound/soc/codecs/Makefile +1 -0

index d7b97ab..cbace60 100644
@@ -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/

sound/soc/codecs/uda1341/Makefile new file +3 -0

new file mode 100644
index 0000000..f9869ba
@@ -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

sound/soc/codecs/uda1341/l3.c new file +30 -0

new file mode 100644
index 0000000..b65b352
@@ -0,0 +1,106 @@
+/*
+ * L3 code
+ *
+ *  Copyright (C) 2008, Christian Pellegrin <[email protected]>
+ *
+ * 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*/

sound/soc/codecs/uda1341/l3.c new file +65 -0

+/*
+ * 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;
+}
+
+

sound/soc/codecs/uda1341/l3.h new file +0 -0

new file mode 100644
index 0000000..2c31e0c

sound/soc/codecs/uda1341/l3.h new file +3 -0

+int l3_write(struct uda1341_platform_data *adap, u8 addr, u8 *data, int len);
+
+#endif

sound/soc/codecs/uda1341/uda1341.c new file +130 -0

new file mode 100644
index 0000000..93cadf0
@@ -0,0 +1,537 @@
+/*
+ * uda1341.c  --  UDA1341 ALSA SoC Codec driver
+ *
+ * Modifications by Christian Pellegrin <[email protected]>
+ *
+ * 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;
+

sound/soc/codecs/uda1341/uda1341.c new file +396 -0

+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 <[email protected]>");
+MODULE_LICENSE("GPL");

sound/soc/codecs/uda1341/uda1341.h new file +47 -0

new file mode 100644
index 0000000..5eccfc4
@@ -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 */

sound/soc/codecs/uda1341/uda1341_platform_data.h new file +0 -0

new file mode 100644
index 0000000..26081cb

sound/soc/codecs/uda1341/uda1341_platform_data.h new file +11 -0

+	int data_hold;
+	int data_setup;
+	int clock_high;
+	int mode_hold;
+	int mode;
+	int mode_setup;
+	void *priv;
+	void (*power) (int);
+};
+
+#endif

sound/soc/s3c24xx/Kconfig +6 -1

index b9f2353..cc8fa5e 100644
@@ -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

sound/soc/s3c24xx/Makefile +2 -0

index 0aa5fb0..155f5a4 100644
@@ -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

sound/soc/s3c24xx/s3c24xx_uda1341.c new file +15 -0

new file mode 100644
index 0000000..fc3c13e
@@ -0,0 +1,314 @@
+/*
+ * Modifications by Christian Pellegrin <[email protected]>
+ *
+ * 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>

sound/soc/s3c24xx/s3c24xx_uda1341.c new file +176 -0

+#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);

sound/soc/s3c24xx/s3c24xx_uda1341.c new file +50 -0

+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);

sound/soc/s3c24xx/s3c24xx_uda1341.c new file +37 -0

+	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 <[email protected]>");
+MODULE_DESCRIPTION("S3C24XX_UDA1341 ALSA SoC audio driver");
+MODULE_LICENSE("GPL");

Quoted 11 lines · On Mon, 10 Nov 2008 20:08:00 +0100 chri <[email protected]> wrote:

Quoted 8 lines · Kristoffer Ericson

On Mon, 10 Nov 2008 13:34:52 +0000

I'm equally interested in this patch, seeing as I
got an hp jornada 7xx with uda1344 waiting.

Have you any plans on creating an 134x seeing as there is very little difference between chipsets?

Quoted 8 lines

Hi,

I had a quick look at the UDA1344 data sheet. When operated in L3-mode it's basically a subset of the UDA1341 so I can do the changes. Anyway I'm not able to test them so could you give me a hand testing? Anyway if you won't use special mixer settings the patch I sent should work since the important configuration registers and the volume are the same for both chips.

It would be good if the driver could be used for both 1341/1344. Some minor ifdef's should sort it out.
Please drop me a mail with the patches (missed the old one) and I'll start testing them them out.

Quoted 7 lines

Signature · 6 lines

-- 
Christian Pellegrin, see http://www.evolware.org/chri/
"Real Programmers don't play tennis, or any other sport which requires
you to change clothes. Mountain climbing is OK, and Real Programmers
wear their climbing boots to work in case a mountain should suddenly
spring up in the middle of the computer room."

Signature · 8 lines

Kristoffer Ericson <[email protected]>



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On Mon, Nov 10, 2008 at 2:34 PM, Mark Brown <[email protected]> wrote:

ack. I was tempted to use pr_debug (and even better dev_dbg) but I went for the old printk method for uniformity with the rest of the drivers.

Quoted 5 lines

sound/soc/codecs/uda1341/uda1341.c line 321

Quoted 2 lines

sound/soc/codecs/uda1341/uda1341.c new file +2 -0

+SOC_SINGLE("DAC Gain switch", UDA1341_STATUS1, 6, 1, 0),
+SOC_SINGLE("ADC Gain switch", UDA1341_STATUS1, 5, 1, 0),

What exactly do these controls do? "Gain switch" sounds wrong - I'd not expect the gain to be a boolean.

they turn on/off an amplifier with 6db gain in the record and playback respectively.

Quoted 7 lines

sound/soc/s3c24xx/s3c24xx_uda1341.c line 61

Quoted 3 lines

sound/soc/s3c24xx/s3c24xx_uda1341.c new file +3 -0

+static int s3c24xx_uda1341_hw_params(struct snd_pcm_substream *substream,
+                                       struct snd_pcm_hw_params *params)
+{

I can follow this function but it'd be nice to see more comments in here. It looks like you could clarify it by iterating over a table of possible clock sources rather than writing each out in code.

unfortunately the 2 clock sources are handled differently: PCLK can be divided, MPLL_in no. So I don't see much convenience in using a (complicated) table

Quoted 6 lines

sound/soc/s3c24xx/s3c24xx_uda1341.c line 150

Quoted 4 lines

sound/soc/s3c24xx/s3c24xx_uda1341.c new file +4 -0

+     ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, clk_source , clk,
+                                             SND_SOC_CLOCK_IN);
+     if (ret < 0)
+             return ret;

You want to run this through scripts/checkpatch.pl.

I already did it

Quoted 8 lines

sound/soc/s3c24xx/s3c24xx_uda1341.c line 221

Quoted 5 lines

sound/soc/s3c24xx/s3c24xx_uda1341.c new file +5 -0

+static void s3c24xx_uda1341_power(int en)
+{
+     if (s3c24xx_uda1341_l3_pins->power)
+             s3c24xx_uda1341_l3_pins->power(en);
+}

This feels like it ought to be initialised conditionally rather than having the check for the pin it.

It's not clear to me what you mean here.

Quoted 7 lines · Mark Brown

sound/soc/s3c24xx/s3c24xx_uda1341.c line 276

Quoted 2 lines

sound/soc/s3c24xx/s3c24xx_uda1341.c new file +2 -0

+   xtal = clk_get(NULL, "xtal");
+   pclk = clk_get(NULL, "pclk");

This should be done in the init function for the device and should really check the return value in case it can't get the clock for some reason. Ideally there'd be a dev there, but I'd need to check if the s3c24xx stuff does that.

And should not be passing a NULL device to clk_get().

example into a doc file please, or point at an implementation of this.

sound/soc/codecs/uda1341/l3.c line 40

Quoted 3 lines

sound/soc/codecs/uda1341/l3.c new file +3 -0

+#define setdat(adap, val)    (adap->setdat(adap, val))
+#define setclk(adap, val)    (adap->setclk(adap, val))
+#define setmode(adap, val)   (adap->setmode(adap, val))

inline functions are better than #define.

Quoted 1 line

+#include "uda1341_platform_data.h"

please find as better name for that file.

Quoted 1 line

+}

do you really want to be busy-waiting the cpu for that long? how about msleep?

Quoted 1 line

+ void (*setmode) (struct uda1341_platform_data *, int);

no spaces ^

find somewhere for these to live that can be included without resorting to knowing the directory layout.

Quoted 1 line

+}

please use the generic gpio layer, the s3c24xx functions are going to be deprecated and removed as soon as possible, probably after 2.6.28.

Quoted 1 line

+ pclk = clk_get(NULL, "pclk");

check for errors. really you should be passing a device for pclk.

Quoted 5 lines

It would be good if the driver could be used for both 1341/1344. Some minor ifdef's should sort it out.
Please drop me a mail with the patches (missed the old one) and I'll start testing them them out.

eww, ifdefs...

If you work with current git you'll see that the other drivers have been converted to pr_debug().

Quoted 3 lines

they turn on/off an amplifier with 6db gain in the record and playback respectively.

Should be "ADC +6dB Switch" or similar then.

A pointer to a table of dividers would handle it. In any case, it needs more comments.

Quoted 3 lines

Quoted 1 line

You want to run this through scripts/checkpatch.pl.

I already did it

I'd have expected checkpatch to catch the extra space you've got after clk_source there...

This function is going to be used in a vtable where IIRC it can be optional but it needs to check the value of power in s3c24xx_uda1341_l3_pins. It feels like you've got too many layers of indirection here and that, for example, the l3_pins power should be assignable directly to the vtable.

Quoted 2 lines · Ben Dooks

find somewhere for these to live that can be included without resorting to knowing the directory layout.

ack, I'll try include/sound

Quoted 6 lines · Ben Dooks

Quoted 5 lines

It would be good if the driver could be used for both 1341/1344. Some minor ifdef's should sort it out.
Please drop me a mail with the patches (missed the old one) and I'll start testing them them out.

Quoted 2 lines

eww, ifdefs...

I'll try to put the common code in a uda_common.c and the do 2 small drivers.

Signature · 5 lines

Christian Pellegrin, see http://www.evolware.org/chri/
"Real Programmers don't play tennis, or any other sport which requires
you to change clothes. Mountain climbing is OK, and Real Programmers
wear their climbing boots to work in case a mountain should suddenly
spring up in the middle of the computer room."

Quoted 8 lines · Ben Dooks

Quoted 5 lines

It would be good if the driver could be used for both 1341/1344. Some minor ifdef's should sort it out.
Please drop me a mail with the patches (missed the old one) and I'll start testing them them out.

eww, ifdefs...

I'll try to put the common code in a uda_common.c and the do 2 small drivers.

Signature · 5 lines

Christian Pellegrin, see http://www.evolware.org/chri/
"Real Programmers don't play tennis, or any other sport which requires
you to change clothes. Mountain climbing is OK, and Real Programmers
wear their climbing boots to work in case a mountain should suddenly
spring up in the middle of the computer room."

On Tue, 11 Nov 2008 15:15:11 +0100

Oh, just wanted to ask, is the L3 code platform specific or universally crunchable?