[PATCH 00/19] Update and clean up on sound/arm/omap/omap-alsa*[c,h] (take #2)

[PATCH 00/19] Update and clean up on sound/arm/omap/omap-alsa*[c,h] (take #2)

[Eduardo Valentin]

From: Eduardo Valentin <eduardo.valentin@indt.org.br>

Hi all,

This patch series cleans up most of the errors and warnings reported
by checkpatch.pl on sound/arm/omap/omap-alsa*[c,h]. Some checks reported
by checkpatch.pl were ignored, such as removing #if 0 codes. Not all were
eligible for remotion.

There is also one patch that updates the alsa driver for H2.
Since tsc2101 i2c driver was updated, this alsa driver stop
working.

Cheers,

---
Eduardo Valentin

Eduardo Valentin (19):
  Update audio driver for H2 board
  Code clean-up for include/asm-arm/arch-omap/omap-alsa.h
  Code clean-up for sound/arm/omap/omap-alsa-aic23.c
  Code clean-up for sound/arm/omap/omap-alsa-aic23.h
  Code clean-up for sound/arm/omap/omap-alsa-aic23-mixer.c
  Code clean-up for sound/arm/omap/omap-alsa-dma.c
  Code clean-up for sound/arm/omap/omap-alsa-dma.h
  Code clean-up for sound/arm/omap/omap-alsa-sx1-mixer.c
  Code clean-up for sound/arm/omap/omap-alsa-sx1-mixer.h
  Code clean-up for sound/arm/omap/omap-alsa-sx1.c
  Code clean-up for sound/arm/omap/omap-alsa-sx1.h
  Code clean-up for sound/arm/omap/omap-alsa-tsc2101-mixer.c
  Code clean-up for sound/arm/omap/omap-alsa-tsc2101-mixer.h
  Code clean-up for sound/arm/omap/omap-alsa-tsc2101.c
  Code clean-up for sound/arm/omap/omap-alsa-tsc2101.h
  Code clean-up for sound/arm/omap/omap-alsa-tsc2102-mixer.c
  Code clean-up for sound/arm/omap/omap-alsa-tsc2102.c
  Code clean-up for sound/arm/omap/omap-alsa-tsc2102.h
  Code clean-up for sound/arm/omap/omap-alsa.c

 arch/arm/mach-omap1/board-h2.c           |   71 ++--
 include/asm-arm/arch-omap/omap-alsa.h    |   29 +-
 sound/arm/omap/omap-alsa-aic23-mixer.c   |  144 ++++---
 sound/arm/omap/omap-alsa-aic23.c         |   59 ++--
 sound/arm/omap/omap-alsa-aic23.h         |   12 +-
 sound/arm/omap/omap-alsa-dma.c           |  171 ++++-----
 sound/arm/omap/omap-alsa-dma.h           |   22 +-
 sound/arm/omap/omap-alsa-sx1-mixer.c     |   22 +-
 sound/arm/omap/omap-alsa-sx1-mixer.h     |    6 +-
 sound/arm/omap/omap-alsa-sx1.c           |   72 +++-
 sound/arm/omap/omap-alsa-sx1.h           |    3 +-
 sound/arm/omap/omap-alsa-tsc2101-mixer.c |  631 ++++++++++++++++--------------
 sound/arm/omap/omap-alsa-tsc2101-mixer.h |   31 +-
 sound/arm/omap/omap-alsa-tsc2101.c       |  320 ++++++++++-----
 sound/arm/omap/omap-alsa-tsc2101.h       |   11 +-
 sound/arm/omap/omap-alsa-tsc2102-mixer.c |   14 +-
 sound/arm/omap/omap-alsa-tsc2102.c       |   70 ++--
 sound/arm/omap/omap-alsa-tsc2102.h       |    2 +-
 sound/arm/omap/omap-alsa.c               |  163 ++++----
 19 files changed, 1056 insertions(+), 797 deletions(-)

[PATCH 01/19] Update audio driver for H2 board

[Eduardo Valentin]

From: Eduardo Valentin <eduardo.valentin@indt.org.br>

This patch updates the audio driver based on tsc2101 chip
for H2 board. tsc2101 driver was updated against new spi
framework, but its audio driver for H2 wasn't.

Signed-off-by: Eduardo Valentin <eduardo.valentin@indt.org.br>
---
 arch/arm/mach-omap1/board-h2.c           |   71 +++++++++++++++---------------
 sound/arm/omap/omap-alsa-tsc2101-mixer.c |    7 ++-
 sound/arm/omap/omap-alsa-tsc2101-mixer.h |    2 +
 sound/arm/omap/omap-alsa-tsc2101.c       |   28 ++++++++++--
 sound/arm/omap/omap-alsa-tsc2101.h       |    5 ++
 5 files changed, 70 insertions(+), 43 deletions(-)

diff --git a/arch/arm/mach-omap1/board-h2.c b/arch/arm/mach-omap1/board-h2.c
index aa1f8af..4f8e6da 100644
--- a/arch/arm/mach-omap1/board-h2.c
+++ b/arch/arm/mach-omap1/board-h2.c
@@ -320,6 +320,40 @@ static void h2_lcd_dev_init(struct spi_device *tsc2101)
 	platform_device_register(&h2_lcd_device);
 }
 
+static struct omap_mcbsp_reg_cfg mcbsp_regs = {
+	.spcr2 = FREE | FRST | GRST | XRST | XINTM(3),
+	.spcr1 = RINTM(3) | RRST,
+	.rcr2  = RPHASE | RFRLEN2(OMAP_MCBSP_WORD_8) |
+			RWDLEN2(OMAP_MCBSP_WORD_16) | RDATDLY(1),
+	.rcr1  = RFRLEN1(OMAP_MCBSP_WORD_8) | RWDLEN1(OMAP_MCBSP_WORD_16),
+	.xcr2  = XPHASE | XFRLEN2(OMAP_MCBSP_WORD_8) |
+			XWDLEN2(OMAP_MCBSP_WORD_16) | XDATDLY(1) | XFIG,
+	.xcr1  = XFRLEN1(OMAP_MCBSP_WORD_8) | XWDLEN1(OMAP_MCBSP_WORD_16),
+	.srgr1 = FWID(15),
+	.srgr2 = GSYNC | CLKSP | FSGM | FPER(31),
+	.pcr0  = CLKXM | CLKRM | FSXP | FSRP | CLKXP | CLKRP,
+	/*.pcr0 = CLKXP | CLKRP,*/        /* mcbsp: slave */
+};
+
+static struct omap_alsa_codec_config alsa_config = {
+	.name                   = "H2 TSC2101",
+	.mcbsp_regs_alsa        = &mcbsp_regs,
+};
+
+static struct platform_device h2_mcbsp1_device = {
+	.name	= "omap_alsa_mcbsp",
+	.id	= 1,
+	.dev = {
+		.platform_data	= &alsa_config,
+	},
+};
+
+static void h2_audio_dev_init(struct spi_device *tsc2101)
+{
+	h2_mcbsp1_device.dev.platform_data = tsc2101;
+	platform_device_register(&h2_mcbsp1_device);
+}
+
 static int h2_tsc2101_init(struct spi_device *spi)
 {
 	int r;
@@ -344,6 +378,7 @@ static int h2_tsc2101_init(struct spi_device *spi)
 	omap_cfg_reg(N15_1610_UWIRE_CS1);
 
 	h2_lcd_dev_init(spi);
+	h2_audio_dev_init(spi);
 
 	return 0;
 err:
@@ -386,48 +421,12 @@ static struct spi_board_info h2_spi_board_info[] __initdata = {
 	},
 };
 
-static struct omap_mcbsp_reg_cfg mcbsp_regs = {
-	.spcr2 = FREE | FRST | GRST | XRST | XINTM(3),
-	.spcr1 = RINTM(3) | RRST,
-	.rcr2  = RPHASE | RFRLEN2(OMAP_MCBSP_WORD_8) |
-                RWDLEN2(OMAP_MCBSP_WORD_16) | RDATDLY(1),
-	.rcr1  = RFRLEN1(OMAP_MCBSP_WORD_8) | RWDLEN1(OMAP_MCBSP_WORD_16),
-	.xcr2  = XPHASE | XFRLEN2(OMAP_MCBSP_WORD_8) |
-                XWDLEN2(OMAP_MCBSP_WORD_16) | XDATDLY(1) | XFIG,
-	.xcr1  = XFRLEN1(OMAP_MCBSP_WORD_8) | XWDLEN1(OMAP_MCBSP_WORD_16),
-	.srgr1 = FWID(15),
-	.srgr2 = GSYNC | CLKSP | FSGM | FPER(31),
-
-	.pcr0  = CLKXM | CLKRM | FSXP | FSRP | CLKXP | CLKRP,
-	/*.pcr0 = CLKXP | CLKRP,*/        /* mcbsp: slave */
-};
-
-static struct omap_alsa_codec_config alsa_config = {
-	.name                   = "H2 TSC2101",
-	.mcbsp_regs_alsa        = &mcbsp_regs,
-	.codec_configure_dev    = NULL, /* tsc2101_configure, */
-	.codec_set_samplerate   = NULL, /* tsc2101_set_samplerate, */
-	.codec_clock_setup      = NULL, /* tsc2101_clock_setup, */
-	.codec_clock_on         = NULL, /* tsc2101_clock_on, */
-	.codec_clock_off        = NULL, /* tsc2101_clock_off, */
-	.get_default_samplerate = NULL, /* tsc2101_get_default_samplerate, */
-};
-
-static struct platform_device h2_mcbsp1_device = {
-	.name	= "omap_alsa_mcbsp",
-	.id	= 1,
-	.dev = {
-		.platform_data	= &alsa_config,
-	},
-};
-
 static struct platform_device *h2_devices[] __initdata = {
 	&h2_nor_device,
 	&h2_nand_device,
 	&h2_smc91x_device,
 	&h2_irda_device,
 	&h2_kp_device,
-	&h2_mcbsp1_device,
 };
 
 static void __init h2_init_smc91x(void)
diff --git a/sound/arm/omap/omap-alsa-tsc2101-mixer.c b/sound/arm/omap/omap-alsa-tsc2101-mixer.c
index 8faba95..09257d7 100644
--- a/sound/arm/omap/omap-alsa-tsc2101-mixer.c
+++ b/sound/arm/omap/omap-alsa-tsc2101-mixer.c
@@ -43,6 +43,7 @@
 #include "omap-alsa-tsc2101.h"
 #include "omap-alsa-tsc2101-mixer.h"
 
+#include <linux/spi/tsc2101.h>
 #include <linux/types.h>
 #include <sound/initval.h>
 #include <sound/control.h>
@@ -68,7 +69,8 @@ static int current_rec_src 		= REC_SRC_SINGLE_ENDED_MICIN_HED;
  */
 inline void omap_tsc2101_audio_write(u8 address, u16 data)
 {
-	omap_tsc2101_write(PAGE2_AUDIO_CODEC_REGISTERS, address, data);
+	tsc2101_write_sync(mcbsp_dev.tsc2101_dev, PAGE2_AUDIO_CODEC_REGISTERS,
+				address, data);
 }
 
 /* 
@@ -76,7 +78,8 @@ inline void omap_tsc2101_audio_write(u8 address, u16 data)
  */
 inline u16 omap_tsc2101_audio_read(u8 address)
 {
-	return (omap_tsc2101_read(PAGE2_AUDIO_CODEC_REGISTERS, address));
+	return (tsc2101_read_sync(mcbsp_dev.tsc2101_dev,
+					PAGE2_AUDIO_CODEC_REGISTERS, address));
 }
 
 /*
diff --git a/sound/arm/omap/omap-alsa-tsc2101-mixer.h b/sound/arm/omap/omap-alsa-tsc2101-mixer.h
index 513614c..b110b36 100644
--- a/sound/arm/omap/omap-alsa-tsc2101-mixer.h
+++ b/sound/arm/omap/omap-alsa-tsc2101-mixer.h
@@ -76,4 +76,6 @@
 
 #define TSC2101_AUDIO_CODEC_REGISTERS_PAGE2     (2)
 
+extern struct mcbsp_dev_info mcbsp_dev;
+
 #endif /*OMAPALSATSC2101MIXER_H_*/
diff --git a/sound/arm/omap/omap-alsa-tsc2101.c b/sound/arm/omap/omap-alsa-tsc2101.c
index 88ed046..1d8adc1 100644
--- a/sound/arm/omap/omap-alsa-tsc2101.c
+++ b/sound/arm/omap/omap-alsa-tsc2101.c
@@ -22,6 +22,7 @@
 #include <linux/soundcard.h>
 #include <linux/platform_device.h>
 #include <linux/clk.h>
+#include <linux/spi/tsc2101.h>
 #include <asm/io.h>
 #include <asm/arch/mcbsp.h>
 
@@ -38,6 +39,8 @@
 #include <asm/arch/omap-alsa.h>
 #include "omap-alsa-tsc2101.h"
 
+struct mcbsp_dev_info mcbsp_dev;
+
 static struct clk *tsc2101_mclk = 0;
 
 //#define DUMP_TSC2101_AUDIO_REGISTERS
@@ -143,7 +146,8 @@ static struct snd_pcm_hardware tsc2101_snd_omap_alsa_capture = {
  */
 inline void tsc2101_audio_write(u8 address, u16 data)
 {
-	omap_tsc2101_write(PAGE2_AUDIO_CODEC_REGISTERS, address, data);
+	tsc2101_write_sync(mcbsp_dev.tsc2101_dev, PAGE2_AUDIO_CODEC_REGISTERS,
+				address, data);
 }
 
 /* 
@@ -151,7 +155,8 @@ inline void tsc2101_audio_write(u8 address, u16 data)
  */
 inline u16 tsc2101_audio_read(u8 address)
 {
-	return (omap_tsc2101_read(PAGE2_AUDIO_CODEC_REGISTERS, address));
+	return (tsc2101_read_sync(mcbsp_dev.tsc2101_dev,
+					PAGE2_AUDIO_CODEC_REGISTERS, address));
 }
 
 #ifdef DUMP_TSC2101_AUDIO_REGISTERS
@@ -341,9 +346,9 @@ int tsc2101_clock_on(void)
 	       err);
 
 	// Now turn the audio on
-	omap_tsc2101_write(PAGE2_AUDIO_CODEC_REGISTERS,
-			TSC2101_CODEC_POWER_CTRL,
-			0x0000);	
+	tsc2101_write_sync(mcbsp_dev.tsc2101_dev, PAGE2_AUDIO_CODEC_REGISTERS,
+				TSC2101_CODEC_POWER_CTRL,
+				0x0000);
 	return 0;	
 }
 
@@ -382,9 +387,22 @@ int tsc2101_get_default_samplerate(void)
 
 static int __devinit snd_omap_alsa_tsc2101_probe(struct platform_device *pdev)
 {
+	struct spi_device *tsc2101;
 	int	ret;
 	struct	omap_alsa_codec_config *codec_cfg;
 	
+	tsc2101 = dev_get_drvdata(&pdev->dev);
+	if (tsc2101 == NULL) {
+		dev_err(&pdev->dev, "no platform data\n");
+		return -ENODEV;
+	}
+	if (strncmp(tsc2101->modalias, "tsc2101", 8) != 0) {
+		dev_err(&pdev->dev, "tsc2101 not found\n");
+		return -EINVAL;
+	}
+	mcbsp_dev.mcbsp_dev = pdev;
+	mcbsp_dev.tsc2101_dev = tsc2101;
+
 	codec_cfg = pdev->dev.platform_data;
 	if (codec_cfg != NULL) {
 		codec_cfg->hw_constraints_rates	= &tsc2101_hw_constraints_rates;
diff --git a/sound/arm/omap/omap-alsa-tsc2101.h b/sound/arm/omap/omap-alsa-tsc2101.h
index 85dccbb..73999a0 100644
--- a/sound/arm/omap/omap-alsa-tsc2101.h
+++ b/sound/arm/omap/omap-alsa-tsc2101.h
@@ -42,6 +42,11 @@
 
 #define PAGE2_AUDIO_CODEC_REGISTERS	(2)
 
+struct mcbsp_dev_info {
+	struct platform_device *mcbsp_dev;
+	struct spi_device *tsc2101_dev;
+};
+
 struct tsc2101_samplerate_reg_info {
 	u16 sample_rate;
 	u8 divisor;
-- 
1.5.5-rc3.GIT

[PATCH 02/19] Code clean-up for include/asm-arm/arch-omap/omap-alsa.h

[Eduardo Valentin]

From: Eduardo Valentin <eduardo.valentin@indt.org.br>

Removed lots of whitespaces and a few errors and
warnings reported by checkpatch.pl.

Signed-off-by: Eduardo Valentin <eduardo.valentin@indt.org.br>
---
 include/asm-arm/arch-omap/omap-alsa.h |   29 +++++++++++++++++++----------
 1 files changed, 19 insertions(+), 10 deletions(-)

diff --git a/include/asm-arm/arch-omap/omap-alsa.h b/include/asm-arm/arch-omap/omap-alsa.h
index 0f8a7d3..9eaa7f2 100644
--- a/include/asm-arm/arch-omap/omap-alsa.h
+++ b/include/asm-arm/arch-omap/omap-alsa.h
@@ -49,15 +49,23 @@
  * Debug functions
  */
 #undef DEBUG
-//#define DEBUG
+/* #define DEBUG */
 
-#define ERR(ARGS...) printk(KERN_ERR "{%s}-ERROR: ", __FUNCTION__);printk(ARGS);
+#define ERR(ARGS...)						\
+	do {							\
+		printk(KERN_ERR "{%s}-ERROR: ", __func__);	\
+		printk(ARGS);					\
+	} while (0)
 
 #ifdef DEBUG
-#define DPRINTK(ARGS...)  printk(KERN_INFO "<%s>: ",__FUNCTION__);printk(ARGS)
-#define ADEBUG() printk("XXX Alsa debug f:%s, l:%d\n", __FUNCTION__, __LINE__)
-#define FN_IN printk(KERN_INFO "[%s]: start\n", __FUNCTION__)
-#define FN_OUT(n) printk(KERN_INFO "[%s]: end(%u)\n",__FUNCTION__, n)
+#define DPRINTK(ARGS...)					\
+	do {							\
+		printk(KERN_INFO "<%s>: ", __func__);		\
+		printk(ARGS);					\
+	} while (0)
+#define ADEBUG() printk("XXX Alsa debug f:%s, l:%d\n", __func__, __LINE__)
+#define FN_IN printk(KERN_INFO "[%s]: start\n", __func__)
+#define FN_OUT(n) printk(KERN_INFO "[%s]: end(%u)\n", __func__, n)
 #else
 #define DPRINTK(ARGS...)	/* nop */
 #define ADEBUG()		/* nop */
@@ -81,12 +89,12 @@ struct audio_stream {
 	char dma_q_count;	/* DMA Channel Q Count */
 	int active:1;		/* we are using this stream for transfer now */
 	int period;		/* current transfer period */
-	int periods;		/* current count of periods registerd in the DMA engine */
+	int periods;		/* current registered periods in DMA engine */
 	spinlock_t dma_lock;	/* for locking in DMA operations */
 	struct snd_pcm_substream *stream;	/* the pcm stream */
 	unsigned linked:1;	/* dma channels linked */
-	int offset;		/* store start position of the last period in the alsa buffer */
-	int (*hw_start)(void);  /* interface to start HW interface, e.g. McBSP */
+	int offset;		/* start position of last period in alsa buf */
+	int (*hw_start)(void);  /* interface to start HW interface, (McBSP) */
 	int (*hw_stop)(void);   /* interface to stop HW interface, e.g. McBSP */
 };
 
@@ -127,7 +135,8 @@ void snd_omap_suspend_mixer(void);
 void snd_omap_resume_mixer(void);
 #endif
 
-int snd_omap_alsa_post_probe(struct platform_device *pdev, struct omap_alsa_codec_config *config);
+int snd_omap_alsa_post_probe(struct platform_device *pdev,
+				struct omap_alsa_codec_config *config);
 int snd_omap_alsa_remove(struct platform_device *pdev);
 #ifdef CONFIG_PM
 int snd_omap_alsa_suspend(struct platform_device *pdev, pm_message_t state);
-- 
1.5.5-rc3.GIT

[PATCH 03/19] Code clean-up for sound/arm/omap/omap-alsa-aic23.c

[Eduardo Valentin]

From: Eduardo Valentin <eduardo.valentin@indt.org.br>

Removed lots of whitespaces and a few errors and
warnings reported by checkpatch.pl.

Signed-off-by: Eduardo Valentin <eduardo.valentin@indt.org.br>
---
 sound/arm/omap/omap-alsa-aic23.c |   59 ++++++++++++++++++++------------------
 sound/arm/omap/omap-alsa-aic23.h |    2 +
 2 files changed, 33 insertions(+), 28 deletions(-)

diff --git a/sound/arm/omap/omap-alsa-aic23.c b/sound/arm/omap/omap-alsa-aic23.c
index 4d67fa3..1e43608 100644
--- a/sound/arm/omap/omap-alsa-aic23.c
+++ b/sound/arm/omap/omap-alsa-aic23.c
@@ -1,6 +1,6 @@
 /*
  * arch/arm/mach-omap1/omap-alsa-aic23.c
- * 
+ *
  * Alsa codec Driver for AIC23 chip on OSK5912 platform board
  *
  * Copyright (C) 2005 Instituto Nokia de Tecnologia - INdT - Manaus Brazil
@@ -8,7 +8,7 @@
  *            {daniel.petrini, david.cohen, anderson.briglia}@indt.org.br
  *
  * Copyright (C) 2006 Mika Laitio <lamikr@cc.jyu.fi>
- * 
+ *
  * Based in former alsa driver for osk and oss driver
  *
  * This program is free software; you can redistribute it and/or modify it
@@ -28,7 +28,7 @@
 #include <asm/arch/omap-alsa.h>
 #include "omap-alsa-aic23.h"
 
-static struct clk *aic23_mclk = 0;
+static struct clk *aic23_mclk;
 
 /* aic23 related */
 static const struct aic23_samplerate_reg_info
@@ -48,7 +48,7 @@ static const struct aic23_samplerate_reg_info
 /*
  * Hardware capabilities
  */
- 
+
  /*
  * DAC USB-mode sampling rates (MCLK = 12 MHz)
  * The rates and rate_reg_into MUST be in the same order
@@ -67,7 +67,7 @@ static struct snd_pcm_hw_constraint_list aic23_hw_constraints_rates = {
 
 static struct snd_pcm_hardware aic23_snd_omap_alsa_playback = {
 	.info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
-		 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID),	
+		 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID),
 	.formats = (SNDRV_PCM_FMTBIT_S16_LE),
 	.rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
 		  SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |
@@ -112,8 +112,6 @@ static struct snd_pcm_hardware aic23_snd_omap_alsa_capture = {
  * codec dependent code.
  */
 
-extern int aic23_write_value(u8 reg, u16 value);
-
 /* TLV320AIC23 is a write only device */
 void audio_aic23_write(u8 address, u16 data)
 {
@@ -172,8 +170,12 @@ inline void aic23_configure(void)
 
 	/* Initialize the AIC23 internal state */
 
-	/* Analog audio path control, DAC selected, delete INSEL_MIC for line in */
-	audio_aic23_write(ANALOG_AUDIO_CONTROL_ADDR, DEFAULT_ANALOG_AUDIO_CONTROL);
+	/*
+	 * Analog audio path control, DAC selected,
+	 * delete INSEL_MIC for line-in
+	 */
+	audio_aic23_write(ANALOG_AUDIO_CONTROL_ADDR,
+				DEFAULT_ANALOG_AUDIO_CONTROL);
 
 	/* Digital audio path control, de-emphasis control 44.1kHz */
 	audio_aic23_write(DIGITAL_AUDIO_CONTROL_ADDR, DEEMP_44K);
@@ -191,11 +193,11 @@ inline void aic23_configure(void)
 }
 
 /*
- *  Omap MCBSP clock configuration and Power Management
- *  
- *  Here we have some functions that allows clock to be enabled and
- *   disabled only when needed. Besides doing clock configuration 
- *   it allows turn on/turn off audio when necessary. 
+ *  OMAP MCBSP clock configuration and Power Management
+ *
+ *  Here we have some functions that allow clock to be enabled and
+ *   disabled only when needed. Besides doing clock configuration
+ *   it allows turn on/turn off audio when necessary.
  */
 /*
  * Do clock framework mclk search
@@ -207,7 +209,7 @@ void aic23_clock_setup(void)
 
 /*
  * Do some sanity check, set clock rate, starts it and
- *  turn codec audio on 
+ *  turn codec audio on
  */
 int aic23_clock_on(void)
 {
@@ -236,9 +238,9 @@ int aic23_clock_on(void)
 	       clk_get_usecount(aic23_mclk));
 
 	/* Now turn the audio on */
-	audio_aic23_write(POWER_DOWN_CONTROL_ADDR, 
+	audio_aic23_write(POWER_DOWN_CONTROL_ADDR,
 			  ~DEVICE_POWER_OFF & ~OUT_OFF & ~DAC_OFF &
-			  ~ADC_OFF & ~MIC_OFF & ~LINE_OFF);	
+			  ~ADC_OFF & ~MIC_OFF & ~LINE_OFF);
 	return 0;
 }
 
@@ -248,7 +250,7 @@ int aic23_clock_on(void)
  */
 int aic23_clock_off(void)
 {
-	if  (clk_get_usecount(aic23_mclk) > 0) { 
+	if (clk_get_usecount(aic23_mclk) > 0) {
 		if (clk_get_rate(aic23_mclk) != CODEC_CLOCK) {
 			printk(KERN_WARNING
 			       "MCLK for audio should be %d Hz. But is %d Hz\n",
@@ -258,10 +260,10 @@ int aic23_clock_off(void)
 
 		clk_disable(aic23_mclk);
 	}
-	
+
 	audio_aic23_write(POWER_DOWN_CONTROL_ADDR,
 			  DEVICE_POWER_OFF | OUT_OFF | DAC_OFF |
-			  ADC_OFF | MIC_OFF | LINE_OFF);	
+			  ADC_OFF | MIC_OFF | LINE_OFF);
 	return 0;
 }
 
@@ -274,21 +276,22 @@ static int __devinit snd_omap_alsa_aic23_probe(struct platform_device *pdev)
 {
 	int	ret;
 	struct	omap_alsa_codec_config *codec_cfg;
-	
+
 	codec_cfg = pdev->dev.platform_data;
 	if (codec_cfg != NULL) {
 		codec_cfg->hw_constraints_rates	= &aic23_hw_constraints_rates;
-		codec_cfg->snd_omap_alsa_playback  = &aic23_snd_omap_alsa_playback;
-		codec_cfg->snd_omap_alsa_capture  = &aic23_snd_omap_alsa_capture;		
+		codec_cfg->snd_omap_alsa_playback =
+						&aic23_snd_omap_alsa_playback;
+		codec_cfg->snd_omap_alsa_capture = &aic23_snd_omap_alsa_capture;
 		codec_cfg->codec_configure_dev	= aic23_configure;
 		codec_cfg->codec_set_samplerate	= aic23_set_samplerate;
 		codec_cfg->codec_clock_setup	= aic23_clock_setup;
 		codec_cfg->codec_clock_on	= aic23_clock_on;
 		codec_cfg->codec_clock_off	= aic23_clock_off;
-		codec_cfg->get_default_samplerate = aic23_get_default_samplerate;
+		codec_cfg->get_default_samplerate =
+						aic23_get_default_samplerate;
 		ret	= snd_omap_alsa_post_probe(pdev, codec_cfg);
-	}
-	else
+	} else
 		ret = -ENODEV;
 	return ret;
 }
@@ -306,7 +309,7 @@ static struct platform_driver omap_alsa_driver = {
 static int __init omap_alsa_aic23_init(void)
 {
 	int err;
-	
+
 	ADEBUG();
 	err = platform_driver_register(&omap_alsa_driver);
 
@@ -316,7 +319,7 @@ static int __init omap_alsa_aic23_init(void)
 static void __exit omap_alsa_aic23_exit(void)
 {
 	ADEBUG();
-	
+
 	platform_driver_unregister(&omap_alsa_driver);
 }
 
diff --git a/sound/arm/omap/omap-alsa-aic23.h b/sound/arm/omap/omap-alsa-aic23.h
index e2e2d59..8b11a69 100644
--- a/sound/arm/omap/omap-alsa-aic23.h
+++ b/sound/arm/omap/omap-alsa-aic23.h
@@ -67,6 +67,8 @@ struct aic23_samplerate_reg_info {
 	u8 divider;		/* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */
 };
 
+extern int aic23_write_value(u8 reg, u16 value);
+
 /*
  * Defines codec specific functions pointers that can be used from the 
  * common omap-alse base driver for all omap codecs. (tsc2101 and aic23)
-- 
1.5.5-rc3.GIT

[PATCH 04/19] Code clean-up for sound/arm/omap/omap-alsa-aic23.h

[Eduardo Valentin]

From: Eduardo Valentin <eduardo.valentin@indt.org.br>

Removed lots of whitespaces and a few errors and
warnings reported by checkpatch.pl.

Signed-off-by: Eduardo Valentin <eduardo.valentin@indt.org.br>
---
 sound/arm/omap/omap-alsa-aic23.h |    9 +++++----
 1 files changed, 5 insertions(+), 4 deletions(-)

diff --git a/sound/arm/omap/omap-alsa-aic23.h b/sound/arm/omap/omap-alsa-aic23.h
index 8b11a69..66dc56b 100644
--- a/sound/arm/omap/omap-alsa-aic23.h
+++ b/sound/arm/omap/omap-alsa-aic23.h
@@ -1,6 +1,6 @@
 /*
  * sound/arm/omap-alsa-aic23.h
- * 
+ *
  * Alsa Driver for AIC23 codec on OSK5912 platform board
  *
  * Copyright (C) 2005 Instituto Nokia de Tecnologia - INdT - Manaus Brazil
@@ -59,7 +59,8 @@
 #define SIDETONE_12			0x080
 #define SIDETONE_18			0x0c0
 
-#define DEFAULT_ANALOG_AUDIO_CONTROL  DAC_SELECTED | STE_ENABLED | BYPASS_ON | INSEL_MIC | MICB_20DB
+#define DEFAULT_ANALOG_AUDIO_CONTROL  (DAC_SELECTED | STE_ENABLED | \
+					BYPASS_ON | INSEL_MIC | MICB_20DB)
 
 struct aic23_samplerate_reg_info {
 	u32 sample_rate;
@@ -70,8 +71,8 @@ struct aic23_samplerate_reg_info {
 extern int aic23_write_value(u8 reg, u16 value);
 
 /*
- * Defines codec specific functions pointers that can be used from the 
- * common omap-alse base driver for all omap codecs. (tsc2101 and aic23)
+ * Defines codec specific function pointers that can be used from the
+ * common omap-alsa base driver for all omap codecs. (tsc2101 and aic23)
  */
 void define_codec_functions(struct omap_alsa_codec_config *codec_config);
 inline void aic23_configure(void);
-- 
1.5.5-rc3.GIT

[PATCH 05/19] Code clean-up for sound/arm/omap/omap-alsa-aic23-mixer.c

[Eduardo Valentin]

From: Eduardo Valentin <eduardo.valentin@indt.org.br>

Removed lots of whitespaces and a few errors and
warnings reported by checkpatch.pl.

Signed-off-by: Eduardo Valentin <eduardo.valentin@indt.org.br>
---
 sound/arm/omap/omap-alsa-aic23-mixer.c |  144 ++++++++++++++++++-------------
 sound/arm/omap/omap-alsa-aic23.h       |    1 +
 2 files changed, 85 insertions(+), 60 deletions(-)

diff --git a/sound/arm/omap/omap-alsa-aic23-mixer.c b/sound/arm/omap/omap-alsa-aic23-mixer.c
index ce55824..7f06582 100644
--- a/sound/arm/omap/omap-alsa-aic23-mixer.c
+++ b/sound/arm/omap/omap-alsa-aic23-mixer.c
@@ -46,7 +46,9 @@
 #include <sound/initval.h>
 #include <sound/control.h>
 
-MODULE_AUTHOR("David Cohen, Daniel Petrini - INdT");
+MODULE_AUTHOR("David Cohen");
+MODULE_AUTHOR("Daniel Petrini");
+
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("OMAP Alsa mixer driver for ALSA");
 
@@ -55,9 +57,8 @@ MODULE_DESCRIPTION("OMAP Alsa mixer driver for ALSA");
  */
 
 /* Codec AIC23 */
-#if defined(CONFIG_SENSORS_TLV320AIC23) || defined (CONFIG_SENSORS_TLV320AIC23_MODULE)
-
-extern void audio_aic23_write(u8, u16);
+#if defined(CONFIG_SENSORS_TLV320AIC23) || \
+	defined(CONFIG_SENSORS_TLV320AIC23_MODULE)
 
 #define MIXER_NAME		     "Mixer AIC23"
 #define SND_OMAP_WRITE(reg, val)     audio_aic23_write(reg, val)
@@ -73,7 +74,8 @@ extern void audio_aic23_write(u8, u16);
 	.info = snd_omap_info_bool, \
 	.get = snd_omap_get_bool, \
 	.put = snd_omap_put_bool, \
-	.private_value = reg | (reg_index << 8) | (invert << 10) | (mask << 12) \
+	.private_value = reg | (reg_index << 8) | (invert << 10) | \
+				(mask << 12) \
 }
 
 #define OMAP_MUX(xname, reg, reg_index, mask) \
@@ -94,7 +96,8 @@ extern void audio_aic23_write(u8, u16);
 	.info = snd_omap_info_single, \
 	.get = snd_omap_get_single, \
 	.put = snd_omap_put_single, \
-	.private_value = reg | (reg_val << 8) | (reg_index << 16) | (mask << 18) \
+	.private_value = reg | (reg_val << 8) | (reg_index << 16) |\
+				(mask << 18) \
 }
 
 #define OMAP_DOUBLE(xname, xindex, left_reg, right_reg, reg_index, mask) \
@@ -105,7 +108,8 @@ extern void audio_aic23_write(u8, u16);
 	.info = snd_omap_info_double, \
 	.get = snd_omap_get_double, \
 	.put = snd_omap_put_double, \
-	.private_value = left_reg | (right_reg << 8) | (reg_index << 16) | (mask << 18) \
+	.private_value = left_reg | (right_reg << 8) | (reg_index << 16) | \
+				(mask << 18) \
 }
 
 /* Local Registers */
@@ -141,7 +145,7 @@ u16 snd_sidetone[6] = {
 /* Begin Bool Functions */
 
 static int snd_omap_info_bool(struct snd_kcontrol *kcontrol,
-			      struct snd_ctl_elem_info * uinfo)
+				struct snd_ctl_elem_info *uinfo)
 {
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
 	uinfo->count = 1;
@@ -151,8 +155,8 @@ static int snd_omap_info_bool(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static int snd_omap_get_bool(struct snd_kcontrol * kcontrol,
-			     struct snd_ctl_elem_value * ucontrol)
+static int snd_omap_get_bool(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
 {
 	int mic_index = (kcontrol->private_value >> 8) & 0x03;
 	u16 mask = (kcontrol->private_value >> 12) & 0xff;
@@ -168,8 +172,8 @@ static int snd_omap_get_bool(struct snd_kcontrol * kcontrol,
 	return 0;
 }
 
-static int snd_omap_put_bool(struct snd_kcontrol * kcontrol,
-			     struct snd_ctl_elem_value * ucontrol)
+static int snd_omap_put_bool(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
 {
 	int mic_index = (kcontrol->private_value >> 8) & 0x03;
 	u16 mask = (kcontrol->private_value >> 12) & 0xff;
@@ -199,7 +203,7 @@ static int snd_omap_put_bool(struct snd_kcontrol * kcontrol,
 /* Begin Mux Functions */
 
 static int snd_omap_info_mux(struct snd_kcontrol *kcontrol,
-			     struct snd_ctl_elem_info * uinfo)
+				struct snd_ctl_elem_info *uinfo)
 {
 	/* Mic = 0
 	 * Line = 1 */
@@ -218,20 +222,20 @@ static int snd_omap_info_mux(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static int snd_omap_get_mux(struct snd_kcontrol * kcontrol,
-			    struct snd_ctl_elem_value * ucontrol)
+static int snd_omap_get_mux(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
 {
 	u16 mask = (kcontrol->private_value >> 10) & 0xff;
-	int mux_index = (kcontrol->private_value >> 8) & 0x03;
+	int mux_idx = (kcontrol->private_value >> 8) & 0x03;
 
 	ucontrol->value.enumerated.item[0] =
-		(omap_regs[mux_index].l_reg & mask) ? 0 /* Mic */ : 1 /* Line */;
+		(omap_regs[mux_idx].l_reg & mask) ? 0 /* Mic */ : 1 /* Line */;
 
 	return 0;
 }
 
-static int snd_omap_put_mux(struct snd_kcontrol * kcontrol,
-			    struct snd_ctl_elem_value * ucontrol)
+static int snd_omap_put_mux(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
 {
 	u16 reg = kcontrol->private_value & 0xff;
 	u16 mask = (kcontrol->private_value >> 10) & 0xff;
@@ -254,7 +258,7 @@ static int snd_omap_put_mux(struct snd_kcontrol * kcontrol,
 /* Begin Single Functions */
 
 static int snd_omap_info_single(struct snd_kcontrol *kcontrol,
-				struct snd_ctl_elem_info * uinfo)
+				struct snd_ctl_elem_info *uinfo)
 {
 	int mask = (kcontrol->private_value >> 18) & 0xff;
 	int reg_val = (kcontrol->private_value >> 8) & 0xff;
@@ -268,8 +272,8 @@ static int snd_omap_info_single(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static int snd_omap_get_single(struct snd_kcontrol * kcontrol,
-			       struct snd_ctl_elem_value * ucontrol)
+static int snd_omap_get_single(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
 {
 	u16 reg_val = (kcontrol->private_value >> 8) & 0xff;
 
@@ -278,8 +282,8 @@ static int snd_omap_get_single(struct snd_kcontrol * kcontrol,
 	return 0;
 }
 
-static int snd_omap_put_single(struct snd_kcontrol * kcontrol,
-			       struct snd_ctl_elem_value * ucontrol)
+static int snd_omap_put_single(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
 {
 	u16 reg_index = (kcontrol->private_value >> 16) & 0x03;
 	u16 mask = (kcontrol->private_value >> 18) & 0x1ff;
@@ -311,10 +315,12 @@ static int snd_omap_put_single(struct snd_kcontrol * kcontrol,
 /* Begin Double Functions */
 
 static int snd_omap_info_double(struct snd_kcontrol *kcontrol,
-				struct snd_ctl_elem_info * uinfo)
+				struct snd_ctl_elem_info *uinfo)
 {
-	/* mask == 0 : Switch
-	 * mask != 0 : Volume */
+	/*
+	 * mask == 0 : Switch
+	 * mask != 0 : Volume
+	 */
 	int mask = (kcontrol->private_value >> 18) & 0xff;
 
 	uinfo->type = mask ? SNDRV_CTL_ELEM_TYPE_INTEGER :
@@ -326,11 +332,13 @@ static int snd_omap_info_double(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static int snd_omap_get_double(struct snd_kcontrol * kcontrol,
-			       struct snd_ctl_elem_value * ucontrol)
+static int snd_omap_get_double(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
 {
-	/* mask == 0 : Switch
-	 * mask != 0 : Volume */
+	/*
+	 * mask == 0 : Switch
+	 * mask != 0 : Volume
+	 */
 	int mask = (kcontrol->private_value >> 18) & 0xff;
 	int vol_index = (kcontrol->private_value >> 16) & 0x03;
 
@@ -346,8 +354,8 @@ static int snd_omap_get_double(struct snd_kcontrol * kcontrol,
 	return 0;
 }
 
-static int snd_omap_put_double(struct snd_kcontrol * kcontrol,
-			       struct snd_ctl_elem_value * ucontrol)
+static int snd_omap_put_double(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
 {
 	/* mask == 0 : Switch
 	 * mask != 0 : Volume */
@@ -371,8 +379,10 @@ static int snd_omap_put_double(struct snd_kcontrol * kcontrol,
 		omap_regs[vol_index].sw = ucontrol->value.integer.value[0];
 	} else {
 		/* Volume */
-		if ((omap_regs[vol_index].l_reg != (ucontrol->value.integer.value[0] & mask)) ||
-			(omap_regs[vol_index].r_reg != (ucontrol->value.integer.value[1] & mask))) {
+		if ((omap_regs[vol_index].l_reg !=
+				(ucontrol->value.integer.value[0] & mask)) ||
+			(omap_regs[vol_index].r_reg !=
+				(ucontrol->value.integer.value[1] & mask))) {
 			changed = 1;
 
 			omap_regs[vol_index].l_reg &= ~mask;
@@ -383,11 +393,12 @@ static int snd_omap_put_double(struct snd_kcontrol * kcontrol,
 				(ucontrol->value.integer.value[1] & mask);
 			if (omap_regs[vol_index].sw) {
 				/* write to registers only if sw is actived */
-				SND_OMAP_WRITE(left_reg, omap_regs[vol_index].l_reg);
-				SND_OMAP_WRITE(right_reg, omap_regs[vol_index].r_reg);
+				SND_OMAP_WRITE(left_reg,
+						omap_regs[vol_index].l_reg);
+				SND_OMAP_WRITE(right_reg,
+						omap_regs[vol_index].r_reg);
 			}
-		}
-		else {
+		} else {
 			changed = 0;
 		}
 	}
@@ -398,20 +409,27 @@ static int snd_omap_put_double(struct snd_kcontrol * kcontrol,
 /* End Double Functions */
 
 static struct snd_kcontrol_new snd_omap_controls[] = {
-	OMAP_DOUBLE("PCM Playback Switch", 0, LEFT_CHANNEL_VOLUME_ADDR, RIGHT_CHANNEL_VOLUME_ADDR,
-		     PCM_INDEX, 0x00),
-	OMAP_DOUBLE("PCM Playback Volume", 0, LEFT_CHANNEL_VOLUME_ADDR, RIGHT_CHANNEL_VOLUME_ADDR,
-		     PCM_INDEX, OUTPUT_VOLUME_MASK),
-	OMAP_BOOL("Line Playback Switch", 0, ANALOG_AUDIO_CONTROL_ADDR, AAC_INDEX, BYPASS_ON, 0),
-	OMAP_DOUBLE("Line Capture Switch", 0, LEFT_LINE_VOLUME_ADDR, RIGHT_LINE_VOLUME_ADDR,
-		     LINE_INDEX, 0x00),
-	OMAP_DOUBLE("Line Capture Volume", 0, LEFT_LINE_VOLUME_ADDR, RIGHT_LINE_VOLUME_ADDR,
-			 LINE_INDEX, INPUT_VOLUME_MASK),
-	OMAP_BOOL("Mic Playback Switch", 0, ANALOG_AUDIO_CONTROL_ADDR, AAC_INDEX, STE_ENABLED, 0),
-	OMAP_SINGLE("Mic Playback Volume", 0, ANALOG_AUDIO_CONTROL_ADDR, AAC_INDEX, 5, SIDETONE_MASK),
-	OMAP_BOOL("Mic Capture Switch", 0, ANALOG_AUDIO_CONTROL_ADDR, AAC_INDEX, MICM_MUTED, 1),
-	OMAP_BOOL("Mic Booster Playback Switch", 0, ANALOG_AUDIO_CONTROL_ADDR, AAC_INDEX, MICB_20DB, 0),
-	OMAP_MUX("Capture Source", ANALOG_AUDIO_CONTROL_ADDR, AAC_INDEX, INSEL_MIC),
+	OMAP_DOUBLE("PCM Playback Switch", 0, LEFT_CHANNEL_VOLUME_ADDR,
+			RIGHT_CHANNEL_VOLUME_ADDR, PCM_INDEX, 0x00),
+	OMAP_DOUBLE("PCM Playback Volume", 0, LEFT_CHANNEL_VOLUME_ADDR,
+			RIGHT_CHANNEL_VOLUME_ADDR, PCM_INDEX,
+			OUTPUT_VOLUME_MASK),
+	OMAP_BOOL("Line Playback Switch", 0, ANALOG_AUDIO_CONTROL_ADDR,
+			AAC_INDEX, BYPASS_ON, 0),
+	OMAP_DOUBLE("Line Capture Switch", 0, LEFT_LINE_VOLUME_ADDR,
+			RIGHT_LINE_VOLUME_ADDR, LINE_INDEX, 0x00),
+	OMAP_DOUBLE("Line Capture Volume", 0, LEFT_LINE_VOLUME_ADDR,
+			RIGHT_LINE_VOLUME_ADDR, LINE_INDEX, INPUT_VOLUME_MASK),
+	OMAP_BOOL("Mic Playback Switch", 0, ANALOG_AUDIO_CONTROL_ADDR,
+			AAC_INDEX, STE_ENABLED, 0),
+	OMAP_SINGLE("Mic Playback Volume", 0, ANALOG_AUDIO_CONTROL_ADDR,
+			AAC_INDEX, 5, SIDETONE_MASK),
+	OMAP_BOOL("Mic Capture Switch", 0, ANALOG_AUDIO_CONTROL_ADDR,
+			AAC_INDEX, MICM_MUTED, 1),
+	OMAP_BOOL("Mic Booster Playback Switch", 0, ANALOG_AUDIO_CONTROL_ADDR,
+			AAC_INDEX, MICB_20DB, 0),
+	OMAP_MUX("Capture Source", ANALOG_AUDIO_CONTROL_ADDR, AAC_INDEX,
+			INSEL_MIC),
 };
 
 #ifdef CONFIG_PM
@@ -447,8 +465,10 @@ void snd_omap_resume_mixer(void)
 	omap_regs[PCM_INDEX].l_reg = omap_pm_regs[PCM_INDEX].l_reg;
 	omap_regs[PCM_INDEX].r_reg = omap_pm_regs[PCM_INDEX].r_reg;
 	omap_regs[PCM_INDEX].sw = omap_pm_regs[PCM_INDEX].sw;
-	SND_OMAP_WRITE(LEFT_CHANNEL_VOLUME_ADDR, omap_pm_regs[PCM_INDEX].l_reg);
-	SND_OMAP_WRITE(RIGHT_CHANNEL_VOLUME_ADDR, omap_pm_regs[PCM_INDEX].r_reg);
+	SND_OMAP_WRITE(LEFT_CHANNEL_VOLUME_ADDR,
+			omap_pm_regs[PCM_INDEX].l_reg);
+	SND_OMAP_WRITE(RIGHT_CHANNEL_VOLUME_ADDR,
+			omap_pm_regs[PCM_INDEX].r_reg);
 }
 #endif
 
@@ -460,8 +480,10 @@ void snd_omap_init_mixer(void)
 	omap_regs[LINE_INDEX].l_reg = DEFAULT_INPUT_VOLUME & INPUT_VOLUME_MASK;
 	omap_regs[LINE_INDEX].r_reg = DEFAULT_INPUT_VOLUME & INPUT_VOLUME_MASK;
 	omap_regs[LINE_INDEX].sw = 0;
-	SND_OMAP_WRITE(LEFT_LINE_VOLUME_ADDR, DEFAULT_INPUT_VOLUME & INPUT_VOLUME_MASK);
-	SND_OMAP_WRITE(RIGHT_LINE_VOLUME_ADDR, DEFAULT_INPUT_VOLUME & INPUT_VOLUME_MASK);
+	SND_OMAP_WRITE(LEFT_LINE_VOLUME_ADDR,
+			DEFAULT_INPUT_VOLUME & INPUT_VOLUME_MASK);
+	SND_OMAP_WRITE(RIGHT_LINE_VOLUME_ADDR,
+			DEFAULT_INPUT_VOLUME & INPUT_VOLUME_MASK);
 
 	/* Analog Audio Control's default values */
 	omap_regs[AAC_INDEX].l_reg = DEFAULT_ANALOG_AUDIO_CONTROL;
@@ -490,10 +512,12 @@ int snd_omap_mixer(struct snd_card_omap_codec *chip)
 	strcpy(card->mixername, MIXER_NAME);
 
 	/* Registering alsa mixer controls */
-	for (idx = 0; idx < ARRAY_SIZE(snd_omap_controls); idx++)
-		if ((err = snd_ctl_add(card,
-			snd_ctl_new1(&snd_omap_controls[idx], chip))) < 0)
+	for (idx = 0; idx < ARRAY_SIZE(snd_omap_controls); idx++) {
+		err = snd_ctl_add(card,
+			snd_ctl_new1(&snd_omap_controls[idx], chip));
+		if (err < 0)
 			return err;
+	}
 
 	return 0;
 }
diff --git a/sound/arm/omap/omap-alsa-aic23.h b/sound/arm/omap/omap-alsa-aic23.h
index 66dc56b..e7dfc15 100644
--- a/sound/arm/omap/omap-alsa-aic23.h
+++ b/sound/arm/omap/omap-alsa-aic23.h
@@ -74,6 +74,7 @@ extern int aic23_write_value(u8 reg, u16 value);
  * Defines codec specific function pointers that can be used from the
  * common omap-alsa base driver for all omap codecs. (tsc2101 and aic23)
  */
+void audio_aic23_write(u8 address, u16 data);
 void define_codec_functions(struct omap_alsa_codec_config *codec_config);
 inline void aic23_configure(void);
 void aic23_set_samplerate(long rate);
-- 
1.5.5-rc3.GIT

[PATCH 06/19] Code clean-up for sound/arm/omap/omap-alsa-dma.c

[Eduardo Valentin]

From: Eduardo Valentin <eduardo.valentin@indt.org.br>

Removed lots of whitespaces and a few errors and
warnings reported by checkpatch.pl.

Signed-off-by: Eduardo Valentin <eduardo.valentin@indt.org.br>
---
 sound/arm/omap/omap-alsa-dma.c |  171 +++++++++++++++++++--------------------
 1 files changed, 83 insertions(+), 88 deletions(-)

diff --git a/sound/arm/omap/omap-alsa-dma.c b/sound/arm/omap/omap-alsa-dma.c
index 00359cb..3f9ff68 100644
--- a/sound/arm/omap/omap-alsa-dma.c
+++ b/sound/arm/omap/omap-alsa-dma.c
@@ -6,7 +6,7 @@
  * Copyright (C) 2006 Mika Laitio <lamikr@cc.jyu.fi>
  *
  * Copyright (C) 2005 Instituto Nokia de Tecnologia - INdT - Manaus Brazil
- * 
+ *
  * Copyright (C) 2004 Texas Instruments, Inc.
  *
  * Copyright (C) 2000, 2001 Nicolas Pitre <nico@cam.org>
@@ -21,24 +21,30 @@
  *
  * History:
  *
- * 2004-06-07	Sriram Kannan	- Created new file from omap_audio_dma_intfc.c. This file
- * 				  will contain only the DMA interface and buffer handling of OMAP
- * 				  audio driver.
+ * 2004-06-07	Sriram Kannan	- Created new file from omap_audio_dma_intfc.c.
+ * 				This file will contain only the DMA interface
+ * 				and buffer handling of OMAP audio driver.
+ *
+ * 2004-06-22	Sriram Kannan	- removed legacy code (auto-init). Self-linking
+ * 				of DMA logical channel.
  *
- * 2004-06-22	Sriram Kannan	- removed legacy code (auto-init). Self-linking of DMA logical channel.
+ * 2004-08-12   Nishanth Menon  - Modified to integrate Audio requirements on
+ * 				1610, 1710 platforms
  *
- * 2004-08-12   Nishanth Menon  - Modified to integrate Audio requirements on 1610,1710 platforms
+ * 2004-11-01   Nishanth Menon  - 16xx platform code base modified to support
+ * 				multi channel chaining.
  *
- * 2004-11-01   Nishanth Menon  - 16xx platform code base modified to support multi channel chaining.
+ * 2004-12-15   Nishanth Menon  - Improved 16xx platform channel logic
+ * 				introduced - tasklets, queue handling updated
  *
- * 2004-12-15   Nishanth Menon  - Improved 16xx platform channel logic introduced - tasklets, queue handling updated
- * 
- * 2005-07-19	INdT Kernel Team - Alsa port. Creation of new file omap-alsa-dma.c based in
- * 				   omap-audio-dma-intfc.c oss file. Support for aic23 codec.
- * 				   Removal of buffer handling (Alsa does that), modifications
- *	in dma handling and port to alsa structures.
+ * 2005-07-19	INdT Kernel Team - Alsa port. Creation of new file
+ * 				omap-alsa-dma.c based in omap-audio-dma-intfc.c
+ * 				oss file. Support for aic23 codec. Removal of
+ * 				buffer handling (Alsa does that), modifications
+ * 				in dma handling and port to alsa structures.
  *
- * 2005-12-18   Dirk Behme      - Added L/R Channel Interchange fix as proposed by Ajaya Babu
+ * 2005-12-18   Dirk Behme      - Added L/R Channel Interchange fix as proposed
+ * 				by Ajaya Babu
  */
 
 #include <linux/module.h>
@@ -56,24 +62,21 @@
 #include <linux/sysrq.h>
 #include <linux/interrupt.h>
 #include <linux/dma-mapping.h>
+#include <linux/io.h>
+#include <linux/uaccess.h>
 
-#include <asm/uaccess.h>
-#include <asm/io.h>
 #include <asm/hardware.h>
 #include <asm/semaphore.h>
-
 #include <asm/arch/dma.h>
-#include "omap-alsa-dma.h"
-
 #include <asm/arch/mcbsp.h>
-
 #include <asm/arch/omap-alsa.h>
 
-#undef DEBUG
+#include "omap-alsa-dma.h"
 
-#define ERR(ARGS...) printk(KERN_ERR "{%s}-ERROR: ", __FUNCTION__);printk(ARGS);
+#undef DEBUG
 
-/* Channel Queue Handling macros
+/*
+ * Channel Queue Handling macros
  * tail always points to the current free entry
  * Head always points to the current entry being used
  * end is either head or tail
@@ -83,13 +86,21 @@
 #define AUDIO_QUEUE_FULL(s) (nr_linked_channels == s->dma_q_count)
 #define AUDIO_QUEUE_LAST(s) (1 == s->dma_q_count)
 #define AUDIO_QUEUE_EMPTY(s) (0 == s->dma_q_count)
-#define __AUDIO_INCREMENT_QUEUE(end) ((end)=((end)+1) % nr_linked_channels)
-#define AUDIO_INCREMENT_HEAD(s) __AUDIO_INCREMENT_QUEUE(s->dma_q_head); s->dma_q_count--;
-#define AUDIO_INCREMENT_TAIL(s) __AUDIO_INCREMENT_QUEUE(s->dma_q_tail); s->dma_q_count++;
+#define __AUDIO_INCREMENT_QUEUE(end) ((end) = ((end)+1) % nr_linked_channels)
+#define AUDIO_INCREMENT_HEAD(s) 			\
+	do {						\
+		__AUDIO_INCREMENT_QUEUE(s->dma_q_head);	\
+		s->dma_q_count--;			\
+	} while (0)
+#define AUDIO_INCREMENT_TAIL(s)				\
+	do {						\
+		__AUDIO_INCREMENT_QUEUE(s->dma_q_tail);	\
+		s->dma_q_count++;			\
+	} while (0)
 
 /* DMA buffer fragmentation sizes */
 #define MAX_DMA_SIZE		 0x1000000 /* todo: sync with alsa */
-//#define CUT_DMA_SIZE		 0x1000
+/* #define CUT_DMA_SIZE		 0x1000 */
 /* TODO: To be moved to more appropriate location */
 #define DCSR_ERROR           0x3
 #define DCSR_END_BLOCK       (1 << 5)
@@ -100,7 +111,7 @@
 #define DCCR_EN              (1 << 7)
 #define DCCR_AI              (1 << 8)
 #define DCCR_REPEAT          (1 << 9)
-/* if 0 the channel works in 3.1 compatible mode*/
+/* if 0 the channel works in 3.1 compatible mode */
 #define DCCR_N31COMP         (1 << 10)
 #define DCCR_EP              (1 << 11)
 #define DCCR_SRC_AMODE_BIT   12
@@ -112,26 +123,22 @@
 #define AMODE_SINGLE_INDEX   0x2
 #define AMODE_DOUBLE_INDEX   0x3
 
-/**************************** DATA STRUCTURES *****************************************/
-
-static spinlock_t dma_list_lock = SPIN_LOCK_UNLOCKED;
-
+/* Data structures */
+DEFINE_SPINLOCK(dma_list_lock);
 static char nr_linked_channels = 1;
 
-/*********************************** MODULE SPECIFIC FUNCTIONS ***********************/
+/* Module specific functions */
 
 static void sound_dma_irq_handler(int lch, u16 ch_status, void *data);
 static int audio_set_dma_params_play(int channel, dma_addr_t dma_ptr,
 				     u_int dma_size);
 static int audio_set_dma_params_capture(int channel, dma_addr_t dma_ptr,
 					u_int dma_size);
-static int audio_start_dma_chain(struct audio_stream * s);
+static int audio_start_dma_chain(struct audio_stream *s);
 
-/***************************************************************************************
- *
+/*
  * DMA channel requests
- *
- **************************************************************************************/
+ */
 static void omap_sound_dma_link_lch(void *data)
 {
 
@@ -166,8 +173,7 @@ int omap_request_alsa_sound_dma(int device_id, const char *device_name,
 		return -EPERM;
 	}
 	/* Try allocate memory for the num channels */
-	*channels =
-	    (int *) kmalloc(sizeof(int) * nr_linked_channels, GFP_KERNEL);
+	*channels = kmalloc(sizeof(int) * nr_linked_channels, GFP_KERNEL);
 	chan = *channels;
 	if (NULL == chan) {
 		ERR("No Memory for channel allocs!\n");
@@ -176,18 +182,19 @@ int omap_request_alsa_sound_dma(int device_id, const char *device_name,
 	}
 	spin_lock(&dma_list_lock);
 	for (i = 0; i < nr_linked_channels; i++) {
-		err = omap_request_dma(device_id, 
+		err = omap_request_dma(device_id,
 				device_name,
-				sound_dma_irq_handler, 
+				sound_dma_irq_handler,
 				data,
 				&chan[i]);
 
 		/* Handle Failure condition here */
 		if (err < 0) {
 			int j;
-			for (j = 0; j < i; j++) {
+
+			for (j = 0; j < i; j++)
 				omap_free_dma(chan[j]);
-			}
+
 			spin_unlock(&dma_list_lock);
 			kfree(chan);
 			*channels = NULL;
@@ -206,12 +213,11 @@ int omap_request_alsa_sound_dma(int device_id, const char *device_name,
 	FN_OUT(0);
 	return 0;
 }
+EXPORT_SYMBOL(omap_request_alsa_sound_dma);
 
-/***************************************************************************************
- *
+/*
  * DMA channel requests Freeing
- *
- **************************************************************************************/
+ */
 static void omap_sound_dma_unlink_lch(void *data)
 {
 	struct audio_stream *s = (struct audio_stream *)data;
@@ -238,7 +244,7 @@ int omap_free_alsa_sound_dma(void *data, int **channels)
 {
 	int i;
 	int *chan = NULL;
-	
+
 	FN_IN;
 	if (unlikely(NULL == channels)) {
 		BUG();
@@ -262,17 +268,16 @@ int omap_free_alsa_sound_dma(void *data, int **channels)
 	FN_OUT(0);
 	return 0;
 }
+EXPORT_SYMBOL(omap_free_alsa_sound_dma);
 
-/***************************************************************************************
- *
+/*
  * Stop all the DMA channels of the stream
- *
- **************************************************************************************/
+ */
 void omap_stop_alsa_sound_dma(struct audio_stream *s)
 {
 	int *chan = s->lch;
 	int i;
-	
+
 	FN_IN;
 	if (unlikely(NULL == chan)) {
 		BUG();
@@ -286,31 +291,30 @@ void omap_stop_alsa_sound_dma(struct audio_stream *s)
 	FN_OUT(0);
 	return;
 }
-/***************************************************************************************
- *
+EXPORT_SYMBOL(omap_stop_alsa_sound_dma);
+
+/*
  * Clear any pending transfers
- *
- **************************************************************************************/
-void omap_clear_alsa_sound_dma(struct audio_stream * s)
+ */
+void omap_clear_alsa_sound_dma(struct audio_stream *s)
 {
 	FN_IN;
 	omap_clear_dma(s->lch[s->dma_q_head]);
 	FN_OUT(0);
 	return;
 }
+EXPORT_SYMBOL(omap_clear_alsa_sound_dma);
 
-/***************************************************************************************
- *
+/*
  * DMA related functions
- *
- **************************************************************************************/
+ */
 static int audio_set_dma_params_play(int channel, dma_addr_t dma_ptr,
 				     u_int dma_size)
 {
 	int dt = 0x1;		/* data type 16 */
 	int cen = 32;		/* Stereo */
 	int cfn = dma_size / (2 * cen);
-	
+
 	FN_IN;
 	omap_set_dma_dest_params(channel, 0x05, 0x00,
 				 (OMAP1510_MCBSP1_BASE + 0x06),
@@ -328,7 +332,7 @@ static int audio_set_dma_params_capture(int channel, dma_addr_t dma_ptr,
 	int dt = 0x1;		/* data type 16 */
 	int cen = 32;		/* stereo */
 	int cfn = dma_size / (2 * cen);
-	
+
 	FN_IN;
 	omap_set_dma_src_params(channel, 0x05, 0x00,
 				(OMAP1510_MCBSP1_BASE + 0x02),
@@ -344,7 +348,7 @@ static int audio_start_dma_chain(struct audio_stream *s)
 	int channel = s->lch[s->dma_q_head];
 	FN_IN;
 	if (!s->started) {
-	 	s->hw_stop();	   /* stops McBSP Interface */
+		s->hw_stop();	   /* stops McBSP Interface */
 		omap_start_dma(channel);
 		s->started = 1;
 		s->hw_start();	   /* start McBSP interface */
@@ -355,12 +359,13 @@ static int audio_start_dma_chain(struct audio_stream *s)
 	return 0;
 }
 
-/* Start DMA -
+/*
+ * Start DMA -
  * Do the initial set of work to initialize all the channels as required.
  * We shall then initate a transfer
  */
-int omap_start_alsa_sound_dma(struct audio_stream *s, 
-			dma_addr_t dma_ptr, 
+int omap_start_alsa_sound_dma(struct audio_stream *s,
+			dma_addr_t dma_ptr,
 			u_int dma_size)
 {
 	int ret = -EPERM;
@@ -372,13 +377,9 @@ int omap_start_alsa_sound_dma(struct audio_stream *s,
 		    MAX_DMA_SIZE);
 		return -EOVERFLOW;
 	}
-	//if (AUDIO_QUEUE_FULL(s)) {
-	//      ret = -2;
-	//      goto sound_out;
-	//}
 
 	if (s->stream_id == SNDRV_PCM_STREAM_PLAYBACK) {
-		/*playback */
+		/* playback */
 		ret =
 		    audio_set_dma_params_play(s->lch[s->dma_q_tail],
 					      dma_ptr, dma_size);
@@ -393,17 +394,18 @@ int omap_start_alsa_sound_dma(struct audio_stream *s,
 	}
 	AUDIO_INCREMENT_TAIL(s);
 	ret = audio_start_dma_chain(s);
-	if (ret) {
+	if (ret)
 		ERR("dma start failed");
-	}
-      sound_out:
+
+sound_out:
 	FN_OUT(ret);
 	return ret;
 
 }
+EXPORT_SYMBOL(omap_start_alsa_sound_dma);
 
-/* 
- * ISRs have to be short and smart.. 
+/*
+ * ISRs have to be short and smart..
  * Here we call alsa handling, after some error checking
  */
 static void sound_dma_irq_handler(int sound_curr_lch, u16 ch_status,
@@ -413,9 +415,7 @@ static void sound_dma_irq_handler(int sound_curr_lch, u16 ch_status,
 	struct audio_stream *s = (struct audio_stream *) data;
 	FN_IN;
 
-	/*
-	 * some register checkings
-	 */ 
+	/* some register checking */
 	DPRINTK("lch=%d,status=0x%x, dma_status=%d, data=%p\n",
 		sound_curr_lch, ch_status, dma_status, data);
 
@@ -426,7 +426,7 @@ static void sound_dma_irq_handler(int sound_curr_lch, u16 ch_status,
 		return;
 	}
 
-	if (ch_status & DCSR_END_BLOCK) 
+	if (ch_status & DCSR_END_BLOCK)
 		callback_omap_alsa_sound_dma(s);
 	FN_OUT(0);
 	return;
@@ -436,8 +436,3 @@ MODULE_AUTHOR("Texas Instruments");
 MODULE_DESCRIPTION("Common DMA handling for Audio driver on OMAP processors");
 MODULE_LICENSE("GPL");
 
-EXPORT_SYMBOL(omap_start_alsa_sound_dma);
-EXPORT_SYMBOL(omap_clear_alsa_sound_dma);
-EXPORT_SYMBOL(omap_request_alsa_sound_dma);
-EXPORT_SYMBOL(omap_free_alsa_sound_dma);
-EXPORT_SYMBOL(omap_stop_alsa_sound_dma);
-- 
1.5.5-rc3.GIT

[PATCH 07/19] Code clean-up for sound/arm/omap/omap-alsa-dma.h

[Eduardo Valentin]

From: Eduardo Valentin <eduardo.valentin@indt.org.br>

Removed lots of whitespaces and a few errors and
warnings reported by checkpatch.pl.

Signed-off-by: Eduardo Valentin <eduardo.valentin@indt.org.br>
---
 sound/arm/omap/omap-alsa-dma.h |   22 +++++++++++-----------
 1 files changed, 11 insertions(+), 11 deletions(-)

diff --git a/sound/arm/omap/omap-alsa-dma.h b/sound/arm/omap/omap-alsa-dma.h
index 1cecc8a..2f0e4e8 100644
--- a/sound/arm/omap/omap-alsa-dma.h
+++ b/sound/arm/omap/omap-alsa-dma.h
@@ -1,4 +1,4 @@
-/*  
+/*
  * linux/sound/arm/omap/omap-alsa-dma.h
  *
  * Common audio DMA handling for the OMAP processors
@@ -6,7 +6,7 @@
  * Copyright (C) 2006 Mika Laitio <lamikr@cc.jyu.fi>
  *
  * Copyright (C) 2005 Instituto Nokia de Tecnologia - INdT - Manaus Brazil
- * 
+ *
  * Copyright (C) 2004 Texas Instruments, Inc.
  *
  * Copyright (C) 2000, 2001 Nicolas Pitre <nico@cam.org>
@@ -21,8 +21,9 @@
  *
  * History:
  *
- * 
- * 2004/08/12  Nishanth Menon - Modified to integrate Audio requirements on 1610,1710 platforms
+ *
+ * 2004/08/12  Nishanth Menon - Modified to integrate Audio requirements on
+ * 				1610, 1710 platforms
  *
  * 2005/07/25  INdT Kernel Team - Renamed to omap-alsa-dma.h. Ported to Alsa.
  */
@@ -30,23 +31,22 @@
 #ifndef __OMAP_AUDIO_ALSA_DMA_H
 #define __OMAP_AUDIO_ALSA_DMA_H
 
-/************************** INCLUDES *************************************/
-
 #include <asm/arch/omap-alsa.h>
 
-/************************** GLOBAL DATA STRUCTURES *********************************/
+/* Global data structures */
 
 typedef void (*dma_callback_t) (int lch, u16 ch_status, void *data);
 
-/**************** ARCH SPECIFIC FUNCIONS *******************************************/
+/* arch specific functions */
 
-void omap_clear_alsa_sound_dma(struct audio_stream * s);
+void omap_clear_alsa_sound_dma(struct audio_stream *s);
 
 int omap_request_alsa_sound_dma(int device_id, const char *device_name,
-			   void *data, int **channels);
+				void *data, int **channels);
 int omap_free_alsa_sound_dma(void *data, int **channels);
 
-int omap_start_alsa_sound_dma(struct audio_stream *s, dma_addr_t dma_ptr,  u_int dma_size);
+int omap_start_alsa_sound_dma(struct audio_stream *s, dma_addr_t dma_ptr,
+				u_int dma_size);
 
 void omap_stop_alsa_sound_dma(struct audio_stream *s);
 
-- 
1.5.5-rc3.GIT

[PATCH 08/19] Code clean-up for sound/arm/omap/omap-alsa-sx1-mixer.c

[Eduardo Valentin]

From: Eduardo Valentin <eduardo.valentin@indt.org.br>

Removed lots of whitespaces and a few errors and
warnings reported by checkpatch.pl.

Signed-off-by: Eduardo Valentin <eduardo.valentin@indt.org.br>
---
 sound/arm/omap/omap-alsa-sx1-mixer.c |   22 +++++++++++-----------
 1 files changed, 11 insertions(+), 11 deletions(-)

diff --git a/sound/arm/omap/omap-alsa-sx1-mixer.c b/sound/arm/omap/omap-alsa-sx1-mixer.c
index 631b080..d0c3322 100644
--- a/sound/arm/omap/omap-alsa-sx1-mixer.c
+++ b/sound/arm/omap/omap-alsa-sx1-mixer.c
@@ -111,9 +111,9 @@ static int pcm_playback_target_info(struct snd_kcontrol *kcontrol,
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
 	uinfo->count = 1;
 	uinfo->value.enumerated.items = PLAYBACK_TARGET_COUNT;
-	if (uinfo->value.enumerated.item > PLAYBACK_TARGET_COUNT - 1) {
+	if (uinfo->value.enumerated.item > PLAYBACK_TARGET_COUNT - 1)
 		uinfo->value.enumerated.item = PLAYBACK_TARGET_COUNT - 1;
-	}
+
 	strcpy(uinfo->value.enumerated.name,
 			texts[uinfo->value.enumerated.item]);
 	return 0;
@@ -210,14 +210,14 @@ static int headset_playback_volume_info(struct snd_kcontrol *kcontrol,
 }
 
 static int headset_playback_volume_get(struct snd_kcontrol *kcontrol,
-						struct snd_ctl_elem_value *ucontrol)
+					struct snd_ctl_elem_value *ucontrol)
 {
 	ucontrol->value.integer.value[0]	= current_volume;
 	return 0;
 }
 
 static int headset_playback_volume_put(struct snd_kcontrol *kcontrol,
-						struct snd_ctl_elem_value *ucontrol)
+					struct snd_ctl_elem_value *ucontrol)
 {
 	return set_mixer_volume(ucontrol->value.integer.value[0]);
 }
@@ -233,14 +233,14 @@ static int headset_playback_switch_info(struct snd_kcontrol *kcontrol,
 }
 
 static int headset_playback_switch_get(struct snd_kcontrol *kcontrol,
-						struct snd_ctl_elem_value *ucontrol)
+					struct snd_ctl_elem_value *ucontrol)
 {
 	ucontrol->value.integer.value[0] = 1;
 	return 0;
 }
 
 static int headset_playback_switch_put(struct snd_kcontrol *kcontrol,
-						struct snd_ctl_elem_value *ucontrol)
+					struct snd_ctl_elem_value *ucontrol)
 {
 	/* mute/unmute headset */
 #if 0
@@ -262,14 +262,14 @@ static int fmradio_playback_volume_info(struct snd_kcontrol *kcontrol,
 }
 
 static int fmradio_playback_volume_get(struct snd_kcontrol *kcontrol,
-						struct snd_ctl_elem_value *ucontrol)
+					struct snd_ctl_elem_value *ucontrol)
 {
 	ucontrol->value.integer.value[0] = current_fm_volume;
 	return 0;
 }
 
 static int fmradio_playback_volume_put(struct snd_kcontrol *kcontrol,
-						struct snd_ctl_elem_value *ucontrol)
+					struct snd_ctl_elem_value *ucontrol)
 {
 	int ret = current_fm_volume != ucontrol->value.integer.value[0];
 	int i;
@@ -291,14 +291,14 @@ static int fmradio_playback_switch_info(struct snd_kcontrol *kcontrol,
 }
 
 static int fmradio_playback_switch_get(struct snd_kcontrol *kcontrol,
-						struct snd_ctl_elem_value *ucontrol)
+					struct snd_ctl_elem_value *ucontrol)
 {
 	ucontrol->value.integer.value[0] = 1;
 	return 0;
 }
 
 static int fmradio_playback_switch_put(struct snd_kcontrol *kcontrol,
-						struct snd_ctl_elem_value *ucontrol)
+					struct snd_ctl_elem_value *ucontrol)
 {
 	/* mute/unmute FM radio */
 	if (ucontrol->value.integer.value[0])
@@ -460,7 +460,7 @@ int snd_omap_mixer(struct snd_card_omap_codec *egold)
 	if (!egold)
 		return -EINVAL;
 
-	for (i=0; i < ARRAY_SIZE(egold_control); i++) {
+	for (i = 0; i < ARRAY_SIZE(egold_control); i++) {
 		err = snd_ctl_add(egold->card,
 				snd_ctl_new1(&egold_control[i], egold->card));
 		if (err < 0)
-- 
1.5.5-rc3.GIT

[PATCH 09/19] Code clean-up for sound/arm/omap/omap-alsa-sx1-mixer.h

[Eduardo Valentin]

From: Eduardo Valentin <eduardo.valentin@indt.org.br>

Removed lots of whitespaces and a few errors and
warnings reported by checkpatch.pl.

Signed-off-by: Eduardo Valentin <eduardo.valentin@indt.org.br>
---
 sound/arm/omap/omap-alsa-sx1-mixer.h |    6 ++++--
 1 files changed, 4 insertions(+), 2 deletions(-)

diff --git a/sound/arm/omap/omap-alsa-sx1-mixer.h b/sound/arm/omap/omap-alsa-sx1-mixer.h
index 686b81c..7d55388 100644
--- a/sound/arm/omap/omap-alsa-sx1-mixer.h
+++ b/sound/arm/omap/omap-alsa-sx1-mixer.h
@@ -31,8 +31,10 @@
 /* following are used for register 03h Mixer PGA control bits
    D7-D5 for selecting record source */
 #define REC_SRC_TARGET_COUNT		0x08
-#define REC_SRC_SINGLE_ENDED_MICIN_HED	0x00 /* oss code referred to MIXER_LINE */
-#define REC_SRC_SINGLE_ENDED_MICIN_HND	0x01 /* oss code referred to MIXER_MIC */
+/* OSS code referred to MIXER_LINE */
+#define REC_SRC_SINGLE_ENDED_MICIN_HED	0x00
+/* OSS code referred to MIXER_MIC */
+#define REC_SRC_SINGLE_ENDED_MICIN_HND	0x01
 #define REC_SRC_SINGLE_ENDED_AUX1	0x02
 #define REC_SRC_SINGLE_ENDED_AUX2	0x03
 #define REC_SRC_MICIN_HED_AND_AUX1	0x04
-- 
1.5.5-rc3.GIT

[PATCH 10/19] Code clean-up for sound/arm/omap/omap-alsa-sx1.c

[Eduardo Valentin]

From: Eduardo Valentin <eduardo.valentin@indt.org.br>

Removed lots of whitespaces and a few errors and
warnings reported by checkpatch.pl.

Signed-off-by: Eduardo Valentin <eduardo.valentin@indt.org.br>
---
 sound/arm/omap/omap-alsa-sx1.c |   72 +++++++++++++++++++++++++++------------
 1 files changed, 50 insertions(+), 22 deletions(-)

diff --git a/sound/arm/omap/omap-alsa-sx1.c b/sound/arm/omap/omap-alsa-sx1.c
index c4691cd..efcabff 100644
--- a/sound/arm/omap/omap-alsa-sx1.c
+++ b/sound/arm/omap/omap-alsa-sx1.c
@@ -14,19 +14,18 @@
 #include <linux/soundcard.h>
 #include <linux/platform_device.h>
 #include <linux/clk.h>
-#include <asm/io.h>
-#include <asm/arch/mcbsp.h>
-
+#include <linux/io.h>
+#include <linux/connector.h>
 #include <linux/slab.h>
 #include <linux/pm.h>
+
 #include <asm/arch/dma.h>
 #include <asm/arch/clock.h>
 #include <asm/arch/gpio.h>
-
+#include <asm/arch/mcbsp.h>
 #include <asm/arch/omap-alsa.h>
-#include "omap-alsa-sx1.h"
 
-#include <linux/connector.h>
+#include "omap-alsa-sx1.h"
 
 /* Connector implementation */
 static struct cb_id cn_sx1snd_id = { CN_IDX_SX1SND, CN_VAL_SX1SND };
@@ -36,9 +35,10 @@ static void cn_sx1snd_callback(void *data)
 {
 	struct cn_msg *msg = (struct cn_msg *)data;
 
-	printk("%s: %lu: idx=%x, val=%x, seq=%u, ack=%u, len=%d: %s.\n",
-			__func__, jiffies, msg->id.idx, msg->id.val,
-			msg->seq, msg->ack, msg->len, (char *)msg->data);
+	printk(KERN_INFO
+		"%s: %lu: idx=%x, val=%x, seq=%u, ack=%u, len=%d: %s.\n",
+		__func__, jiffies, msg->id.idx, msg->id.val,
+		msg->seq, msg->ack, msg->len, (char *)msg->data);
 }
 
 /* Send IPC message to sound server */
@@ -63,7 +63,7 @@ int cn_sx1snd_send(unsigned int cmd, unsigned int arg1, unsigned int arg2)
 	memcpy(m + 1, data, m->len);
 
 	err = cn_netlink_send(m, CN_IDX_SX1SND, gfp_any());
-	snd_printd("sent= %02X %02X %02X, err=%d\n", cmd,arg1,arg2,err);
+	snd_printd("sent= %02X %02X %02X, err=%d\n", cmd, arg1, arg2, err);
 	kfree(m);
 
 	if (err == -ESRCH)
@@ -150,15 +150,42 @@ static void egold_set_samplerate(long sample_rate)
 	clkgdv	= CODEC_CLOCK / (sample_rate * (DEFAULT_BITPERSAMPLE * 2 - 1));
 #endif
 	switch (sample_rate) {
-		case 8000:	clkgdv = 71; egold_rate = FRQ_8000; break;
-		case 11025:	clkgdv = 51; egold_rate = FRQ_11025; break;
-		case 12000:	clkgdv = 47; egold_rate = FRQ_12000; break;
-		case 16000:	clkgdv = 35; egold_rate = FRQ_16000; break;
-		case 22050:	clkgdv = 25; egold_rate = FRQ_22050; break;
-		case 24000:	clkgdv = 23; egold_rate = FRQ_24000; break;
-		case 32000:	clkgdv = 17; egold_rate = FRQ_32000; break;
-		case 44100:	clkgdv = 12; egold_rate = FRQ_44100; break;
-		case 48000:	clkgdv = 11; egold_rate = FRQ_48000; break;
+	case 8000:
+		clkgdv = 71;
+		egold_rate = FRQ_8000;
+		break;
+	case 11025:
+		clkgdv = 51;
+		egold_rate = FRQ_11025;
+		break;
+	case 12000:
+		clkgdv = 47;
+		egold_rate = FRQ_12000;
+		break;
+	case 16000:
+		clkgdv = 35;
+		egold_rate = FRQ_16000;
+		break;
+	case 22050:
+		clkgdv = 25;
+		egold_rate = FRQ_22050;
+		break;
+	case 24000:
+		clkgdv = 23;
+		egold_rate = FRQ_24000;
+		break;
+	case 32000:
+		clkgdv = 17;
+		egold_rate = FRQ_32000;
+		break;
+	case 44100:
+		clkgdv = 12;
+		egold_rate = FRQ_44100;
+		break;
+	case 48000:
+		clkgdv = 11;
+		egold_rate = FRQ_48000;
+		break;
 	}
 
 	srgr1 = (FWID(DEFAULT_BITPERSAMPLE - 1) | CLKGDV(clkgdv));
@@ -234,8 +261,8 @@ static int __init snd_omap_alsa_egold_probe(struct platform_device *pdev)
 		return -ENODEV;
 
 	codec_cfg->hw_constraints_rates	= &egold_hw_constraints_rates;
-	codec_cfg->snd_omap_alsa_playback= &egold_snd_omap_alsa_playback;
-	codec_cfg->snd_omap_alsa_capture  = &egold_snd_omap_alsa_capture;
+	codec_cfg->snd_omap_alsa_playback = &egold_snd_omap_alsa_playback;
+	codec_cfg->snd_omap_alsa_capture = &egold_snd_omap_alsa_capture;
 	codec_cfg->codec_configure_dev	= egold_configure;
 	codec_cfg->codec_set_samplerate	= egold_set_samplerate;
 	codec_cfg->codec_clock_setup	= egold_clock_setup;
@@ -262,7 +289,8 @@ static int __init omap_alsa_egold_init(void)
 {
 	int retval;
 
-	retval = cn_add_callback(&cn_sx1snd_id, cn_sx1snd_name, cn_sx1snd_callback);
+	retval = cn_add_callback(&cn_sx1snd_id, cn_sx1snd_name,
+					cn_sx1snd_callback);
 	if (retval)
 		printk(KERN_WARNING "cn_sx1snd failed to register\n");
 	return platform_driver_register(&omap_alsa_driver);
-- 
1.5.5-rc3.GIT

[PATCH 11/19] Code clean-up for sound/arm/omap/omap-alsa-sx1.h

[Eduardo Valentin]

From: Eduardo Valentin <eduardo.valentin@indt.org.br>

Removed lots of whitespaces and a few errors and
warnings reported by checkpatch.pl.

Signed-off-by: Eduardo Valentin <eduardo.valentin@indt.org.br>
---
 sound/arm/omap/omap-alsa-sx1.h |    3 ++-
 1 files changed, 2 insertions(+), 1 deletions(-)

diff --git a/sound/arm/omap/omap-alsa-sx1.h b/sound/arm/omap/omap-alsa-sx1.h
index af7a409..9e11e0a 100644
--- a/sound/arm/omap/omap-alsa-sx1.h
+++ b/sound/arm/omap/omap-alsa-sx1.h
@@ -35,7 +35,8 @@
 #define OSC_EN				2
 
 /* Send IPC message to sound server */
-extern int cn_sx1snd_send(unsigned int cmd, unsigned int arg1, unsigned int arg2);
+extern int cn_sx1snd_send(unsigned int cmd, unsigned int arg1,
+				unsigned int arg2);
 /* cmd for IPC_GROUP_DAC */
 #define DAC_VOLUME_UPDATE		0
 #define DAC_SETAUDIODEVICE		1
-- 
1.5.5-rc3.GIT

[PATCH 12/19] Code clean-up for sound/arm/omap/omap-alsa-tsc2101-mixer.c

[Eduardo Valentin]

[-- Warning: decoded text below may be mangled, UTF-8 assumed --]
[-- Attachment #1: Type: text/plain; charset=unknown-8bit, Size: 43599 bytes --]

From: Eduardo Valentin <eduardo.valentin@indt.org.br>

Removed lots of whitespaces and a few errors and
warnings reported by checkpatch.pl.

Signed-off-by: Eduardo Valentin <eduardo.valentin@indt.org.br>
---
 sound/arm/omap/omap-alsa-tsc2101-mixer.c |  624 ++++++++++++++++--------------
 1 files changed, 342 insertions(+), 282 deletions(-)

diff --git a/sound/arm/omap/omap-alsa-tsc2101-mixer.c b/sound/arm/omap/omap-alsa-tsc2101-mixer.c
index 09257d7..d443342 100644
--- a/sound/arm/omap/omap-alsa-tsc2101-mixer.c
+++ b/sound/arm/omap/omap-alsa-tsc2101-mixer.c
@@ -1,15 +1,15 @@
 /*
  * sound/arm/omap/omap-alsa-tsc2101-mixer.c
- * 
+ *
  * Alsa Driver for TSC2101 codec for OMAP platform boards.
  *
- * Copyright (C) 2005 Mika Laitio <lamikr@cc.jyu.fi> and 
+ * Copyright (C) 2005 Mika Laitio <lamikr@cc.jyu.fi> and
  * 		     Everett Coleman II <gcc80x86@fuzzyneural.net>
  *
  * Board initialization code is based on the code in TSC2101 OSS driver.
  * Copyright (C) 2004 Texas Instruments, Inc.
  * 	Written by Nishanth Menon and Sriram Kannan
- * 
+ *
  * This program is free software; you can redistribute it and/or modify it
  * under the terms of the GNU General Public License as published by the
  * Free Software Foundation; either version 2 of the License, or (at your
@@ -33,13 +33,13 @@
  * History:
  *
  * 2006-03-01   Mika Laitio - Mixer for the tsc2101 driver used in omap boards.
- * 		Can switch between headset and loudspeaker playback, 
+ * 		Can switch between headset and loudspeaker playback,
  * 		mute and unmute dgc, set dgc volume. Record source switch,
- * 		keyclick, buzzer and headset volume and handset volume control 
+ * 		keyclick, buzzer and headset volume and handset volume control
  * 		are still missing.
- * 		
+ *
  */
- 
+
 #include "omap-alsa-tsc2101.h"
 #include "omap-alsa-tsc2101-mixer.h"
 
@@ -48,8 +48,15 @@
 #include <sound/initval.h>
 #include <sound/control.h>
 
-//#define M_DPRINTK(ARGS...)  printk(KERN_INFO "<%s>: ",__FUNCTION__);printk(ARGS)
+#ifdef DEBUG
+#define M_DPRINTK(ARGS...)				\
+	do {						\
+		printk(KERN_INFO "<%s>: ", __func__);	\
+		printk(ARGS);				\
+	} while (0)
+#else
 #define M_DPRINTK(ARGS...)  		/* nop */
+#endif
 
 #define CHECK_BIT(INDX, ARG) (((ARG) & TSC2101_BIT(INDX)) >> INDX)
 #define IS_UNMUTED(INDX, ARG) (((CHECK_BIT(INDX, ARG)) == 0))
@@ -64,7 +71,7 @@
 static int current_playback_target	= PLAYBACK_TARGET_LOUDSPEAKER;
 static int current_rec_src 		= REC_SRC_SINGLE_ENDED_MICIN_HED;
 
-/* 
+/*
  * Simplified write for the tsc2101 audio registers.
  */
 inline void omap_tsc2101_audio_write(u8 address, u16 data)
@@ -73,7 +80,7 @@ inline void omap_tsc2101_audio_write(u8 address, u16 data)
 				address, data);
 }
 
-/* 
+/*
  * Simplified read for the tsc2101 audio registers.
  */
 inline u16 omap_tsc2101_audio_read(u8 address)
@@ -88,8 +95,9 @@ inline u16 omap_tsc2101_audio_read(u8 address)
 static void set_record_source(int val)
 {
 	u16	data;
-	
-	/* Mute Analog Sidetone
+
+	/*
+	 * Mute Analog Sidetone
 	 * Analog sidetone gain db?
 	 * Input selected by MICSEL connected to ADC
 	 */
@@ -98,77 +106,84 @@ static void set_record_source(int val)
 	data	|= MPC_MICSEL(val);
 	data	|= MPC_MICADC;
 	omap_tsc2101_audio_write(TSC2101_MIXER_PGA_CTRL, data);
-	
+
 	current_rec_src	= val;
 }
 
 /*
- * Converts the Alsa mixer volume (0 - 100) to real 
+ * Converts the Alsa mixer volume (0 - 100) to real
  * Digital Gain Control (DGC) value that can be written
  * or read from the TSC2101 registry.
- * 
+ *
  * Note that the number "OUTPUT_VOLUME_MAX" is smaller than OUTPUT_VOLUME_MIN
  * because DGC works as a volume decreaser. (The more bigger value is put
  * to DGC, the more the volume of controlled channel is decreased)
- * 
- * In addition the TCS2101 chip would allow the maximum volume reduction be 63.5 DB
+ *
+ * In addition the TCS2101 chip would allow the maximum
+ * volume reduction be 63.5 DB
  * but according to some tests user can not hear anything with this chip
  * when the volume is set to be less than 25 db.
- * Therefore this function will return a value that means 38.5 db (63.5 db - 25 db) 
+ * Therefore this function will return a value
+ * that means 38.5 db (63.5 db - 25 db)
  * reduction in the channel volume, when mixer is set to 0.
- * For mixer value 100, this will return a value that means 0 db volume reduction.
+ * For mixer value 100, this will return a value that means
+ * 0 db volume reduction.
  * ([mute_left_bit]0000000[mute_right_bit]0000000)
-*/
+ */
 int get_mixer_volume_as_dac_gain_control_volume(int vol)
 {
 	u16 retVal;
 
 	/* Convert 0 -> 100 volume to 0x7F(min) -> y(max) volume range */
-	retVal	= ((vol * OUTPUT_VOLUME_RANGE) / 100) + OUTPUT_VOLUME_MAX;
+	retVal = ((vol * OUTPUT_VOLUME_RANGE) / 100) + OUTPUT_VOLUME_MAX;
 	/* invert the value for getting the proper range 0 min and 100 max */
-	retVal	= OUTPUT_VOLUME_MIN - retVal;
-	
+	retVal = OUTPUT_VOLUME_MIN - retVal;
+
 	return retVal;
 }
 
 /*
- * Converts the Alsa mixer volume (0 - 100) to TSC2101 
+ * Converts the Alsa mixer volume (0 - 100) to TSC2101
  * Digital Gain Control (DGC) volume. Alsa mixer volume 0
  * is converted to value meaning the volume reduction of -38.5 db
  * and Alsa mixer volume 100 is converted to value meaning the
  * reduction of 0 db.
  */
-int set_mixer_volume_as_dac_gain_control_volume(int mixerVolL, int mixerVolR) 
+int set_mixer_volume_as_dac_gain_control_volume(int mixerVolL, int mixerVolR)
 {
 	u16 val;
 	int retVal;
 	int volL;
 	int volR;
-	
-	if ((mixerVolL < 0) || 
+
+	if ((mixerVolL < 0) ||
 	    (mixerVolL > 100) ||
 	    (mixerVolR < 0) ||
 	    (mixerVolR > 100)) {
-		printk(KERN_ERR "Trying a bad mixer volume as dac gain control volume value, left (%d), right (%d)!\n", mixerVolL, mixerVolR);
+		printk(KERN_ERR "Trying a bad mixer volume as dac gain control"
+			" volume value, left (%d), right (%d)!\n", mixerVolL,
+			mixerVolR);
 		return -EPERM;
 	}
-	M_DPRINTK("mixer volume left = %d, right = %d\n", mixerVolL, mixerVolR);	
+	M_DPRINTK("mixer volume left = %d, right = %d\n", mixerVolL, mixerVolR);
 	volL	= get_mixer_volume_as_dac_gain_control_volume(mixerVolL);
 	volR	= get_mixer_volume_as_dac_gain_control_volume(mixerVolR);
-	
+
 	val	= omap_tsc2101_audio_read(TSC2101_DAC_GAIN_CTRL);
 	/* keep the old mute bit settings */
-	val	&= ~(DGC_DALVL(OUTPUT_VOLUME_MIN) | DGC_DARVL(OUTPUT_VOLUME_MIN));
+	val	&= ~(DGC_DALVL(OUTPUT_VOLUME_MIN) |
+			DGC_DARVL(OUTPUT_VOLUME_MIN));
 	val	|= DGC_DALVL(volL) | DGC_DARVL(volR);
 	retVal	= 2;
-	if (retVal) {
+	if (retVal)
 		omap_tsc2101_audio_write(TSC2101_DAC_GAIN_CTRL, val);
-	}
-	M_DPRINTK("to registry: left = %d, right = %d, total = %d\n", DGC_DALVL_EXTRACT(val), DGC_DARVL_EXTRACT(val), val);
+
+	M_DPRINTK("to registry: left = %d, right = %d, total = %d\n",
+			DGC_DALVL_EXTRACT(val), DGC_DARVL_EXTRACT(val), val);
 	return retVal;
 }
 
-/**
+/*
  * If unmuteLeft/unmuteRight == 0  --> mute
  * If unmuteLeft/unmuteRight == 1 --> unmute
  */
@@ -179,15 +194,16 @@ int dac_gain_control_unmute(int unmuteLeft, int unmuteRight)
 
 	count	= 0;
 	val	= omap_tsc2101_audio_read(TSC2101_DAC_GAIN_CTRL);
-	/* in alsa mixer 1 --> on, 0 == off. In tsc2101 registry 1 --> off, 0 --> on
-	 * so if values are same, it's time to change the registry value.
+	/*
+	 * in alsa mixer 1 --> on, 0 == off. In tsc2101 registry 1 --> off,
+	 * 0 --> on so if values are same, it's time to change the registry
+	 * value.
 	 */
 	if (unmuteLeft != IS_UNMUTED(15, val)) {
 		if (unmuteLeft == 0) {
 			/* mute --> turn bit on */
 			val	= val | DGC_DALMU;
-		}
-		else {
+		} else {
 			/* unmute --> turn bit off */
 			val	= val & ~DGC_DALMU;
 		}
@@ -197,69 +213,71 @@ int dac_gain_control_unmute(int unmuteLeft, int unmuteRight)
 		if (unmuteRight == 0) {
 			/* mute --> turn bit on */
 			val	= val | DGC_DARMU;
-		}
-		else {
+		} else {
 			/* unmute --> turn bit off */
 			val	= val & ~DGC_DARMU;
-		}		
+		}
 		count++;
 	} /* R */
 	if (count) {
 		omap_tsc2101_audio_write(TSC2101_DAC_GAIN_CTRL, val);
-		M_DPRINTK("changed value, is_unmuted left = %d, right = %d\n", 
+		M_DPRINTK("changed value, is_unmuted left = %d, right = %d\n",
 			IS_UNMUTED(15, val),
 			IS_UNMUTED(7, val));
 	}
-	return count;	
+	return count;
 }
 
-/**
+/*
  * unmute: 0 --> mute, 1 --> unmute
  * page2RegIndx: Registry index in tsc2101 page2.
- * muteBitIndx: Index number for the bit in registry that indicates whether muted or unmuted.
+ * muteBitIndx: Index number for the bit in registry that indicates whether
+ * muted or unmuted.
  */
 int adc_pga_unmute_control(int unmute, int page2regIndx, int muteBitIndx)
 {
 	int count;
 	u16 val;
-	
+
 	count	= 0;
 	val 	= omap_tsc2101_audio_read(page2regIndx);
-	/* in alsa mixer 1 --> on, 0 == off. In tsc2101 registry 1 --> off, 0 --> on
-	 * so if the values are same, it's time to change the registry value...
+	/*
+	 * in alsa mixer 1 --> on, 0 == off. In tsc2101 registry 1 --> off,
+	 * 0 --> on so if the values are same, it's time to change the
+	 * registry value...
 	 */
 	if (unmute != IS_UNMUTED(muteBitIndx, val)) {
 		if (unmute == 0) {
 			/* mute --> turn bit on */
 			val	= val | TSC2101_BIT(muteBitIndx);
-		}
-		else {
+		} else {
 			/* unmute --> turn bit off */
 			val	= val & ~TSC2101_BIT(muteBitIndx);
 		}
-		M_DPRINTK("changed value, is_unmuted = %d\n", IS_UNMUTED(muteBitIndx, val));
+		M_DPRINTK("changed value, is_unmuted = %d\n",
+				IS_UNMUTED(muteBitIndx, val));
 		count++;
 	}
-	if (count) {
+	if (count)
 		omap_tsc2101_audio_write(page2regIndx, val);
-	}
+
 	return count;
 }
 
 /*
- * Converts the DGC registry value read from the TSC2101 registry to 
+ * Converts the DGC registry value read from the TSC2101 registry to
  * Alsa mixer volume format (0 - 100).
  */
-int get_dac_gain_control_volume_as_mixer_volume(u16 vol) 
+int get_dac_gain_control_volume_as_mixer_volume(u16 vol)
 {
-	u16 retVal;	
+	u16 retVal;
 
 	retVal	= OUTPUT_VOLUME_MIN - vol;
 	retVal	= ((retVal - OUTPUT_VOLUME_MAX) * 100) / OUTPUT_VOLUME_RANGE;
 	/* fix scaling error */
-	if ((retVal > 0) && (retVal < 100)) {
+	if ((retVal > 0) && (retVal < 100))
 		retVal++;
-	}
+
 	return retVal;
 }
 
@@ -267,10 +285,10 @@ int get_dac_gain_control_volume_as_mixer_volume(u16 vol)
  * Converts the headset gain control volume (0 - 63.5 db)
  * to Alsa mixer volume (0 - 100)
  */
-int get_headset_gain_control_volume_as_mixer_volume(u16 registerVal) 
+int get_headset_gain_control_volume_as_mixer_volume(u16 registerVal)
 {
 	u16 retVal;
-	
+
 	retVal	= ((registerVal * 100) / INPUT_VOLUME_RANGE);
 	return retVal;
 }
@@ -279,71 +297,78 @@ int get_headset_gain_control_volume_as_mixer_volume(u16 registerVal)
  * Converts the handset gain control volume (0 - 63.5 db)
  * to Alsa mixer volume (0 - 100)
  */
-int get_handset_gain_control_volume_as_mixer_volume(u16 registerVal) 
+int get_handset_gain_control_volume_as_mixer_volume(u16 registerVal)
 {
 	return get_headset_gain_control_volume_as_mixer_volume(registerVal);
 }
 
 /*
- * Converts the Alsa mixer volume (0 - 100) to 
+ * Converts the Alsa mixer volume (0 - 100) to
  * headset gain control volume (0 - 63.5 db)
  */
-int get_mixer_volume_as_headset_gain_control_volume(u16 mixerVal) 
+int get_mixer_volume_as_headset_gain_control_volume(u16 mixerVal)
 {
 	u16 retVal;
-	
-	retVal	= ((mixerVal * INPUT_VOLUME_RANGE) / 100) + INPUT_VOLUME_MIN;	
+
+	retVal	= ((mixerVal * INPUT_VOLUME_RANGE) / 100) + INPUT_VOLUME_MIN;
 	return retVal;
 }
 
 /*
  * Writes Alsa mixer volume (0 - 100) to TSC2101 headset volume registry in
  * a TSC2101 format. (0 - 63.5 db)
- * In TSC2101 OSS driver this functionality was controlled with "SET_LINE" parameter.
+ * In TSC2101 OSS driver this functionality was controlled with "SET_LINE"
+ * parameter.
  */
-int set_mixer_volume_as_headset_gain_control_volume(int mixerVol) 
+int set_mixer_volume_as_headset_gain_control_volume(int mixerVol)
 {
 	int volume;
 	int retVal;
 	u16 val;
 
 	if (mixerVol < 0 || mixerVol > 100) {
-		M_DPRINTK("Trying a bad headset mixer volume value(%d)!\n", mixerVol);
+		M_DPRINTK("Trying a bad headset mixer volume value(%d)!\n",
+				mixerVol);
 		return -EPERM;
 	}
 	M_DPRINTK("mixer volume = %d\n", mixerVol);
-	/* Convert 0 -> 100 volume to 0x0(min) -> 0x7D(max) volume range */
-	/* NOTE: 0 is minimum volume and not mute */
-	volume	= get_mixer_volume_as_headset_gain_control_volume(mixerVol);	
+	/*
+	 * Convert 0 -> 100 volume to 0x0(min) -> 0x7D(max) volume range
+	 * NOTE: 0 is minimum volume and not mute
+	 */
+	volume	= get_mixer_volume_as_headset_gain_control_volume(mixerVol);
 	val	= omap_tsc2101_audio_read(TSC2101_HEADSET_GAIN_CTRL);
 	/* preserve the old mute settings */
 	val	&= ~(HGC_ADPGA_HED(INPUT_VOLUME_MAX));
 	val	|= HGC_ADPGA_HED(volume);
-	omap_tsc2101_audio_write(TSC2101_HEADSET_GAIN_CTRL, val);	
+	omap_tsc2101_audio_write(TSC2101_HEADSET_GAIN_CTRL, val);
 	retVal	= 1;
-	
-	M_DPRINTK("to registry = %d\n", val);	
+
+	M_DPRINTK("to registry = %d\n", val);
 	return retVal;
 }
 
 /*
  * Writes Alsa mixer volume (0 - 100) to TSC2101 handset volume registry in
  * a TSC2101 format. (0 - 63.5 db)
- * In TSC2101 OSS driver this functionality was controlled with "SET_MIC" parameter.
+ * In TSC2101 OSS driver this functionality was controlled with
+ * "SET_MIC" parameter.
  */
-int set_mixer_volume_as_handset_gain_control_volume(int mixerVol) 
+int set_mixer_volume_as_handset_gain_control_volume(int mixerVol)
 {
 	int volume;
 	int retVal;
-	u16 val;	
+	u16 val;
 
 	if (mixerVol < 0 || mixerVol > 100) {
-		M_DPRINTK("Trying a bad mic mixer volume value(%d)!\n", mixerVol);
+		M_DPRINTK("Trying a bad mic mixer volume value(%d)!\n",
+				mixerVol);
 		return -EPERM;
 	}
 	M_DPRINTK("mixer volume = %d\n", mixerVol);
-	/* Convert 0 -> 100 volume to 0x0(min) -> 0x7D(max) volume range
-	 * NOTE: 0 is minimum volume and not mute 
+	/*
+	 * Convert 0 -> 100 volume to 0x0(min) -> 0x7D(max) volume range
+	 * NOTE: 0 is minimum volume and not mute
 	 */
 	volume	= get_mixer_volume_as_headset_gain_control_volume(mixerVol);
 	val	= omap_tsc2101_audio_read(TSC2101_HANDSET_GAIN_CTRL);
@@ -352,8 +377,8 @@ int set_mixer_volume_as_handset_gain_control_volume(int mixerVol)
 	val	|= HNGC_ADPGA_HND(volume);
 	omap_tsc2101_audio_write(TSC2101_HANDSET_GAIN_CTRL, val);
 	retVal	= 1;
-	
-	M_DPRINTK("to registry = %d\n", val);	
+
+	M_DPRINTK("to registry = %d\n", val);
 	return retVal;
 }
 
@@ -361,27 +386,31 @@ void set_loudspeaker_to_playback_target(void)
 {
 	/* power down SPK1, SPK2 and loudspeaker */
 	omap_tsc2101_audio_write(TSC2101_CODEC_POWER_CTRL,
-			CPC_SP1PWDN | CPC_SP2PWDN | CPC_LDAPWDF);	
-	/* ADC, DAC, Analog Sidetone, cellphone, buzzer softstepping enabled
+			CPC_SP1PWDN | CPC_SP2PWDN | CPC_LDAPWDF);
+	/*
+	 * ADC, DAC, Analog Sidetone, cellphone, buzzer softstepping enabled
 	 * 1dB AGC hysteresis
 	 * MICes bias 2V
 	 */
 	omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_4, AC4_MB_HED(0));
 
-	/* DAC left and right routed to SPK1/SPK2
+	/*
+	 * DAC left and right routed to SPK1/SPK2
 	 * SPK1/SPK2 unmuted
 	 * Keyclicks routed to SPK1/SPK2 */
-	omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_5, 
+	omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_5,
 			AC5_DIFFIN |
 			AC5_DAC2SPK1(3) | AC5_AST2SPK1 | AC5_KCL2SPK1 |
 			AC5_DAC2SPK2(3) | AC5_AST2SPK2 | AC5_KCL2SPK2);
-	
-	/* routing selected to SPK1 goes also to OUT8P/OUT8N. (loudspeaker)
+
+	/*
+	 * routing selected to SPK1 goes also to OUT8P/OUT8N. (loudspeaker)
 	 * analog sidetone routed to loudspeaker
 	 * buzzer pga routed to loudspeaker
 	 * keyclick routing to loudspeaker
 	 * cellphone input routed to loudspeaker
-	 * mic selection (control register 04h/page2) routed to cell phone output (CP_OUT)
+	 * mic selection (control register 04h/page2) routed to cell phone
+	 * output (CP_OUT)
 	 * routing selected for SPK1 goes also to cellphone output (CP_OUT)
 	 * OUT8P/OUT8N (loudspeakers) unmuted (0 = unmuted)
 	 * Cellphone output is not muted (0 = unmuted)
@@ -399,19 +428,23 @@ void set_headphone_to_playback_target(void)
 	/* power down SPK1, SPK2 and loudspeaker */
 	omap_tsc2101_audio_write(TSC2101_CODEC_POWER_CTRL,
 			CPC_SP1PWDN | CPC_SP2PWDN | CPC_LDAPWDF);
-	/* ADC, DAC, Analog Sidetone, cellphone, buzzer softstepping enabled */
-	/* 1dB AGC hysteresis */
-	/* MICes bias 2V */
+	/*
+	 * ADC, DAC, Analog Sidetone, cellphone, buzzer softstepping enabled
+	 * 1dB AGC hysteresis
+	 * MICes bias 2V
+	 */
 	omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_4, AC4_MB_HED(0));
-				
-	/* DAC left and right routed to SPK1/SPK2
+
+	/*
+	 * DAC left and right routed to SPK1/SPK2
 	 * SPK1/SPK2 unmuted
-	 * Keyclicks routed to SPK1/SPK2 */
+	 * Keyclicks routed to SPK1/SPK2
+	 */
 	omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_5,
 			AC5_DAC2SPK1(3) | AC5_AST2SPK1 | AC5_KCL2SPK1 |
 			AC5_DAC2SPK2(3) | AC5_AST2SPK2 | AC5_KCL2SPK2 |
 			AC5_HDSCPTC);
-			
+
 	/* OUT8P/OUT8N muted, CPOUT muted */
 	omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_6,
 			AC6_MUTLSPK | AC6_MUTSPK2 | AC6_LDSCPTC |
@@ -421,45 +454,47 @@ void set_headphone_to_playback_target(void)
 
 void set_telephone_to_playback_target(void)
 {
-	/* 
+	/*
 	 * 0110 1101 0101 1100
-	 * power down MICBIAS_HED, Analog sidetone, SPK2, DAC, 
+	 * power down MICBIAS_HED, Analog sidetone, SPK2, DAC,
 	 * Driver virtual ground, loudspeaker. Values D2-d5 are flags.
-	 */	 
+	 */
 	omap_tsc2101_audio_write(TSC2101_CODEC_POWER_CTRL,
 			CPC_MBIAS_HED | CPC_ASTPWD | CPC_SP2PWDN | CPC_DAPWDN |
 			CPC_VGPWDN | CPC_LSPWDN);
-			
-	/* 
+
+	/*
 	 * 0010 1010 0100 0000
 	 * ADC, DAC, Analog Sidetone, cellphone, buzzer softstepping enabled
 	 * 1dB AGC hysteresis
 	 * MICes bias 2V
 	 */
 	omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_4,
-			AC4_MB_HND | AC4_MB_HED(0) | AC4_AGCHYS(1) | 
+			AC4_MB_HND | AC4_MB_HED(0) | AC4_AGCHYS(1) |
 			AC4_BISTPD | AC4_ASSTPD | AC4_DASTPD);
-	printk("set_telephone_to_playback_target(), TSC2101_AUDIO_CTRL_4 = %d\n", omap_tsc2101_audio_read(TSC2101_AUDIO_CTRL_4));
-			
-	/* 
+	printk(KERN_INFO "set_telephone_to_playback_target(), "
+			"TSC2101_AUDIO_CTRL_4 = %d\n",
+			omap_tsc2101_audio_read(TSC2101_AUDIO_CTRL_4));
+
+	/*
 	 * 1110 0010 0000 0010
 	 * DAC left and right routed to SPK1/SPK2
 	 * SPK1/SPK2 unmuted
 	 * keyclicks routed to SPK1/SPK2
-	 */	 
+	 */
 	omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_5,
-			AC5_DIFFIN | AC5_DAC2SPK1(3) | 
-		  	AC5_CPI2SPK1 | AC5_MUTSPK2);
-	
+			AC5_DIFFIN | AC5_DAC2SPK1(3) |
+			AC5_CPI2SPK1 | AC5_MUTSPK2);
+
 	omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_6,
-			AC6_MIC2CPO | AC6_MUTLSPK | 
+			AC6_MIC2CPO | AC6_MUTLSPK |
 			AC6_LDSCPTC | AC6_VGNDSCPTC | AC6_CAPINTF);
 	current_playback_target	= PLAYBACK_TARGET_CELLPHONE;
 }
 
 /*
  * 1100 0101 1101 0000
- * 
+ *
  * #define MPC_ASTMU           TSC2101_BIT(15)
  * #define MPC_ASTG(ARG)       (((ARG) & 0x7F) << 8)
  * #define MPC_MICSEL(ARG)     (((ARG) & 0x07) << 5)
@@ -470,14 +505,14 @@ void set_telephone_to_playback_target(void)
 static void set_telephone_to_record_source(void)
 {
 	u16	val;
-	
-	/* 
-	 * D0       = 0: 
+
+	/*
+	 * D0       = 0:
 	 * 		--> AGC is off for handset input.
 	 *		--> ADC PGA is controlled by the ADMUT_HDN + ADPGA_HND
 	 *          (D15, D14-D8)
-	 * D4 - D1  = 0000 
-	 * 		--> AGC time constant for handset input, 
+	 * D4 - D1  = 0000
+	 * 		--> AGC time constant for handset input,
 	 * 		attack time = 8 mc, decay time = 100 ms
 	 * D7 - D5  = 000
 	 * 		--> AGC Target gain for handset input = -5.5 db
@@ -486,33 +521,36 @@ static void set_telephone_to_record_source(void)
 	 * D15 		= 0
 	 * 		--> Handset input ON (unmuted)
 	 */
-	val	= 0x3c00;	// 0011 1100 0000 0000 = 60 = 30
+	val	= 0x3c00;	/* 0011 1100 0000 0000 = 60 = 30 */
 	omap_tsc2101_audio_write(TSC2101_HANDSET_GAIN_CTRL, val);
-	
+
 	/*
 	 * D0		= 0
 	 * 		--> AGC is off for headset/Aux input
-	 * 		--> ADC headset/Aux PGA is contoller by ADMUT_HED + ADPGA_HED
+	 * 		--> ADC headset/Aux PGA is contoller by
+	 * 		ADMUT_HED + ADPGA_HED
 	 *          (D15, D14-D8)
-	 * D4 - D1	= 0000 
+	 * D4 - D1	= 0000
 	 * 		--> Agc constant for headset/Aux input,
-	 *      	attack time = 8 mc, decay time = 100 ms      
+	 *      	attack time = 8 mc, decay time = 100 ms
 	 * D7 - D5	= 000
 	 * 		--> AGC target gain for headset input = -5.5 db
 	 * D14 - D8 = 000 0000
 	 * 		--> Adc headset/AUX pga settings = 0 db
 	 * D15		= 1
 	 * 		--> Headset/AUX input muted
-	 * 
+	 *
 	 * Mute headset aux input
 	 */
-	val	= 0x8000;	// 1000 0000 0000 0000
+	val	= 0x8000;	/* 1000 0000 0000 0000 */
 	omap_tsc2101_audio_write(TSC2101_HEADSET_GAIN_CTRL, val);
 	set_record_source(REC_SRC_MICIN_HND_AND_AUX1);
 
-	// hacks start
-	/* D0		= flag, Headset/Aux or handset PGA flag
-	 * 		--> & with 1 (= 1 -->gain applied == pga register settings)
+	/*
+	 * hacks start
+	 * D0		= flag, Headset/Aux or handset PGA flag
+	 * 		--> & with 1 (= 1 -->gain applied == pga
+	 * 		register settings)
 	 * D1		= 0, DAC channel PGA soft stepping control
 	 * 		--> 0.5 db change every WCLK
 	 * D2		= flag, DAC right channel PGA flag
@@ -521,8 +559,8 @@ static void set_telephone_to_record_source(void)
 	 * 		-- > & with 1
 	 * D7 - D4	= 0001, keyclick length
 	 * 		--> 4 periods key clicks
-	 * D10 - D8 = 100, keyclick frequenzy
-	 * 		--> 1 kHz, 
+	 * D10 - D8 = 100, keyclick frequency
+	 * 		--> 1 kHz,
 	 * D11		= 0, Headset/Aux or handset soft stepping control
 	 * 		--> 0,5 db change every WCLK or ADWS
 	 * D14 -D12 = 100, Keyclick applitude control
@@ -531,7 +569,7 @@ static void set_telephone_to_record_source(void)
 	 */
 	val	= omap_tsc2101_audio_read(TSC2101_AUDIO_CTRL_2);
 	val	= val & 0x441d;
-	val	= val | 0x4410;	// D14, D10, D4 bits == 1
+	val	= val | 0x4410;	/* D14, D10, D4 bits == 1 */
 	omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_2, val);
 
 	/*
@@ -543,24 +581,28 @@ static void set_telephone_to_record_source(void)
 	 * 			--> MICBIAS_HND = 2.0 v
 	 * D8 - D7	= 00
 	 * 			--> MICBIAS_HED = 3.3 v
-	 * D10 - D9	= 01, 
+	 * D10 - D9	= 01,
 	 * 			--> Mic AGC hysteric selection = 2 db
-	 * D11		= 1, 
+	 * D11		= 1,
 	 * 			--> Disable buzzer PGA soft stepping
 	 * D12		= 0,
 	 * 			--> Enable CELL phone PGA soft stepping control
 	 * D13		= 1
-	 * 			--> Disable analog sidetone soft stepping control
+	 * 			--> Disable analog sidetone soft
+	 * 			stepping control
 	 * D14		= 0
 	 * 			--> Enable DAC PGA soft stepping control
 	 * D15		= 0,
-	 * 			--> Enable headset/Aux or Handset soft stepping control
+	 * 			--> Enable headset/Aux or Handset soft
+	 * 			stepping control
 	 */
 	val	= omap_tsc2101_audio_read(TSC2101_AUDIO_CTRL_4);
-	val	= val & 0x2a42;	// 0010 1010 0100 0010
-	val	= val | 0x2a40;	// bits D13, D11, D9, D6 == 1
+	val	= val & 0x2a42;	/* 0010 1010 0100 0010 */
+	val	= val | 0x2a40;	/* bits D13, D11, D9, D6 == 1 */
 	omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_4, val);
-	printk("set_telephone_to_record_source(), TSC2101_AUDIO_CTRL_4 = %d\n", omap_tsc2101_audio_read(TSC2101_AUDIO_CTRL_4));
+	printk(KERN_INFO "set_telephone_to_record_source(), "
+			"TSC2101_AUDIO_CTRL_4 = %d\n",
+			omap_tsc2101_audio_read(TSC2101_AUDIO_CTRL_4));
 	/*
 	 * D0		= 0
 	 * 		--> reserved, write always = 0
@@ -579,10 +621,12 @@ static void set_telephone_to_record_source(void)
 	 */
 	val	= omap_tsc2101_audio_read(TSC2101_BUZZER_GAIN_CTRL);
 	val	= val & 0x5dfe;
-	val	= val | 0x5dfe;	// bits, D14, D12, D11, D10, D8, D6, D5,D4,D3,D2
+	/* bits, D14, D12, D11, D10, D8, D6, D5,D4,D3,D2 */
+	val	= val | 0x5dfe;
 	omap_tsc2101_audio_write(TSC2101_BUZZER_GAIN_CTRL, val);
-	
-	/* D6 - D0	= 000 1001
+
+	/*
+	 * D6 - D0	= 000 1001
 	 * 		--> -4.5 db for DAC right channel volume control
 	 * D7		= 1
 	 * 		-->  DAC right channel muted
@@ -591,12 +635,13 @@ static void set_telephone_to_record_source(void)
 	 * D15 		= 1
 	 * 		--> DAC left channel muted
 	 */
-	//val	= omap_tsc2101_audio_read(TSC2101_DAC_GAIN_CTRL);
+	/* val	= omap_tsc2101_audio_read(TSC2101_DAC_GAIN_CTRL); */
 	val	= 0x8989;
-	omap_tsc2101_audio_write(TSC2101_DAC_GAIN_CTRL, val);	
-	
-	/*  0000 0000 0100 0000
-	 * 
+	omap_tsc2101_audio_write(TSC2101_DAC_GAIN_CTRL, val);
+
+	/*
+	 *   0000 0000 0100 0000
+	 *
 	 * D1 - D0	= 0
 	 * 		--> GPIO 1 pin output is three stated
 	 * D2		= 0
@@ -610,18 +655,18 @@ static void set_telephone_to_record_source(void)
 	 * 		--> 8 ms clitch detection
 	 * D8		= reserved, write only 0
 	 * D10 -D9	= 00
-	 * 		--> 16 ms de bouncing programmatitily 
+	 * 		--> 16 ms de-bouncing
 	 *          for glitch detection during headset detection
 	 * D11		= flag for button press
 	 * D12		= flag for headset detection
 	 * D14-D13	= 00
-	 * 		--> type of headset detected = 00 == no stereo headset deected
+	 * 		--> type of headset detected = 00 == no stereo
+	 * 		headset deected
 	 * D15		= 0
 	 * 		--> Disable headset detection
-	 * 
-	 * */
+	 */
 	val	= 0x40;
-	omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_7, val);	
+	omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_7, val);
 }
 
 /*
@@ -637,17 +682,17 @@ u16 get_headset_detected(void)
 	u16	curDetected;
 	u16	curType;
 	u16	curVal;
-	
+
 	curType	= 0;	/* not detected */
 	curVal	= omap_tsc2101_audio_read(TSC2101_AUDIO_CTRL_7);
 	curDetected	= curVal & AC7_HDDETFL;
 	if (curDetected) {
-		printk("headset detected, checking type from %d \n", curVal);
+		printk(KERN_INFO "headset detected, checking type from %d \n",
+			curVal);
 		curType	= ((curVal & 0x6000) >> 13);
-		printk("headset type detected = %d \n", curType);
-	}
-	else {
-		printk("headset not detected\n");
+		printk(KERN_INFO "headset type detected = %d \n", curType);
+	} else {
+		printk(KERN_INFO "headset not detected\n");
 	}
 	return curType;
 }
@@ -657,40 +702,46 @@ void init_playback_targets(void)
 	u16	val;
 
 	set_loudspeaker_to_playback_target();
-	/* Left line input volume control
+	/*
+	 * Left line input volume control
 	 * = SET_LINE in the OSS driver
 	 */
 	set_mixer_volume_as_headset_gain_control_volume(DEFAULT_INPUT_VOLUME);
 
-	/* Set headset to be controllable by handset mixer
+	/*
+	 * Set headset to be controllable by handset mixer
 	 * AGC enable for handset input
 	 * Handset input not muted
 	 */
 	val	= omap_tsc2101_audio_read(TSC2101_HANDSET_GAIN_CTRL);
-	val	= val | HNGC_AGCEN_HND;	
+	val	= val | HNGC_AGCEN_HND;
 	val	= val & ~HNGC_ADMUT_HND;
-	omap_tsc2101_audio_write(TSC2101_HANDSET_GAIN_CTRL, val);	
-			
-	/* mic input volume control
-	 * SET_MIC in the OSS driver 
+	omap_tsc2101_audio_write(TSC2101_HANDSET_GAIN_CTRL, val);
+
+	/*
+	 * mic input volume control
+	 * SET_MIC in the OSS driver
 	 */
 	set_mixer_volume_as_handset_gain_control_volume(DEFAULT_INPUT_VOLUME);
 
-	/* Left/Right headphone channel volume control
+	/*
+	 * Left/Right headphone channel volume control
 	 * Zero-cross detect on
 	 */
-	set_mixer_volume_as_dac_gain_control_volume(DEFAULT_OUTPUT_VOLUME, DEFAULT_OUTPUT_VOLUME);	
+	set_mixer_volume_as_dac_gain_control_volume(DEFAULT_OUTPUT_VOLUME,
+							DEFAULT_OUTPUT_VOLUME);
 	/* unmute */
 	dac_gain_control_unmute(1, 1);
 }
 
 /*
- * Initializes tsc2101 recourd source (to line) and playback target (to loudspeaker)
+ * Initializes tsc2101 recourd source (to line) and playback target
+ * (to loudspeaker)
  */
 void snd_omap_init_mixer(void)
-{	
+{
 	FN_IN;
-	
+
 	/* Headset/Hook switch detect enabled */
 	omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_7, AC7_DETECT);
 
@@ -706,17 +757,17 @@ static int __pcm_playback_target_info(struct snd_kcontrol *kcontrol,
 		struct snd_ctl_elem_info *uinfo)
 {
 	static char *texts[PLAYBACK_TARGET_COUNT] = {
-        	"Loudspeaker", "Headphone", "Cellphone"
+		"Loudspeaker", "Headphone", "Cellphone"
 	};
 
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
 	uinfo->count = 1;
 	uinfo->value.enumerated.items = PLAYBACK_TARGET_COUNT;
-	if (uinfo->value.enumerated.item > PLAYBACK_TARGET_COUNT - 1) {
-        	uinfo->value.enumerated.item = PLAYBACK_TARGET_COUNT - 1;
-	}
+	if (uinfo->value.enumerated.item > PLAYBACK_TARGET_COUNT - 1)
+		uinfo->value.enumerated.item = PLAYBACK_TARGET_COUNT - 1;
+
 	strcpy(uinfo->value.enumerated.name,
-       	texts[uinfo->value.enumerated.item]);
+		texts[uinfo->value.enumerated.item]);
 	return 0;
 }
 
@@ -732,28 +783,26 @@ static int __pcm_playback_target_put(struct snd_kcontrol *kcontrol,
 {
 	int	retVal;
 	int	curVal;
-	
+
 	retVal	= 0;
 	curVal	= ucontrol->value.integer.value[0];
 	if ((curVal >= 0) &&
 	    (curVal < PLAYBACK_TARGET_COUNT) &&
-	    (curVal != current_playback_target)) {		
+	    (curVal != current_playback_target)) {
 		if (curVal == PLAYBACK_TARGET_LOUDSPEAKER) {
 			set_record_source(REC_SRC_SINGLE_ENDED_MICIN_HED);
 			set_loudspeaker_to_playback_target();
-		}
-		else if (curVal == PLAYBACK_TARGET_HEADPHONE) {
+		} else if (curVal == PLAYBACK_TARGET_HEADPHONE) {
 			set_record_source(REC_SRC_SINGLE_ENDED_MICIN_HND);
 			set_headphone_to_playback_target();
-		}
-		else if (curVal == PLAYBACK_TARGET_CELLPHONE) {
+		} else if (curVal == PLAYBACK_TARGET_CELLPHONE) {
 			set_telephone_to_record_source();
 			set_telephone_to_playback_target();
 		}
 		retVal	= 1;
 	}
 	return retVal;
-}	
+}
 
 static int __pcm_playback_volume_info(struct snd_kcontrol *kcontrol,
 		struct snd_ctl_elem_info *uinfo)
@@ -766,16 +815,16 @@ static int __pcm_playback_volume_info(struct snd_kcontrol *kcontrol,
 }
 
 /*
- * Alsa mixer interface function for getting the volume read from the DGC in a 
+ * Alsa mixer interface function for getting the volume read from the DGC in a
  * 0 -100 alsa mixer format.
  */
 static int __pcm_playback_volume_get(struct snd_kcontrol *kcontrol,
 		struct snd_ctl_elem_value *ucontrol)
 {
 	u16 volL;
-	u16 volR;	
+	u16 volR;
 	u16 val;
-	
+
 	val	= omap_tsc2101_audio_read(TSC2101_DAC_GAIN_CTRL);
 	M_DPRINTK("registry value = %d!\n", val);
 	volL	= DGC_DALVL_EXTRACT(val);
@@ -786,19 +835,22 @@ static int __pcm_playback_volume_get(struct snd_kcontrol *kcontrol,
 
 	volL	= get_dac_gain_control_volume_as_mixer_volume(volL);
 	volR	= get_dac_gain_control_volume_as_mixer_volume(volR);
-	
+
 	ucontrol->value.integer.value[0]	= volL; /* L */
 	ucontrol->value.integer.value[1]	= volR; /* R */
-	
-	M_DPRINTK("mixer volume left = %ld, right = %ld\n", ucontrol->value.integer.value[0], ucontrol->value.integer.value[1]);
+
+	M_DPRINTK("mixer volume left = %ld, right = %ld\n",
+			ucontrol->value.integer.value[0],
+			ucontrol->value.integer.value[1]);
 	return 0;
 }
 
 static int __pcm_playback_volume_put(struct snd_kcontrol *kcontrol,
 		struct snd_ctl_elem_value *ucontrol)
 {
-	return set_mixer_volume_as_dac_gain_control_volume(ucontrol->value.integer.value[0], 
-							ucontrol->value.integer.value[1]);
+	return set_mixer_volume_as_dac_gain_control_volume(
+					ucontrol->value.integer.value[0],
+					ucontrol->value.integer.value[1]);
 }
 
 static int __pcm_playback_switch_info(struct snd_kcontrol *kcontrol,
@@ -811,7 +863,7 @@ static int __pcm_playback_switch_info(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-/* 
+/*
  * When DGC_DALMU (bit 15) is 1, the left channel is muted.
  * When DGC_DALMU is 0, left channel is not muted.
  * Same logic apply also for the right channel.
@@ -820,16 +872,16 @@ static int __pcm_playback_switch_get(struct snd_kcontrol *kcontrol,
 		struct snd_ctl_elem_value *ucontrol)
 {
 	u16 val	= omap_tsc2101_audio_read(TSC2101_DAC_GAIN_CTRL);
-	
-	ucontrol->value.integer.value[0]	= IS_UNMUTED(15, val);	// left
-	ucontrol->value.integer.value[1]	= IS_UNMUTED(7, val);	// right
+
+	ucontrol->value.integer.value[0] = IS_UNMUTED(15, val);	/* left */
+	ucontrol->value.integer.value[1] = IS_UNMUTED(7, val); /* right */
 	return 0;
 }
 
 static int __pcm_playback_switch_put(struct snd_kcontrol *kcontrol,
 		struct snd_ctl_elem_value *ucontrol)
 {
-	return dac_gain_control_unmute(ucontrol->value.integer.value[0], 
+	return dac_gain_control_unmute(ucontrol->value.integer.value[0],
 					ucontrol->value.integer.value[1]);
 }
 
@@ -848,7 +900,7 @@ static int __headset_playback_volume_get(struct snd_kcontrol *kcontrol,
 {
 	u16 val;
 	u16 vol;
-	
+
 	val	= omap_tsc2101_audio_read(TSC2101_HEADSET_GAIN_CTRL);
 	M_DPRINTK("registry value = %d\n", val);
 	vol	= HGC_ADPGA_HED_EXTRACT(val);
@@ -856,15 +908,17 @@ static int __headset_playback_volume_get(struct snd_kcontrol *kcontrol,
 
 	vol	= get_headset_gain_control_volume_as_mixer_volume(vol);
 	ucontrol->value.integer.value[0]	= vol;
-	
-	M_DPRINTK("mixer volume returned = %ld\n", ucontrol->value.integer.value[0]);
+
+	M_DPRINTK("mixer volume returned = %ld\n",
+			ucontrol->value.integer.value[0]);
 	return 0;
 }
 
 static int __headset_playback_volume_put(struct snd_kcontrol *kcontrol,
 		struct snd_ctl_elem_value *ucontrol)
 {
-	return set_mixer_volume_as_headset_gain_control_volume(ucontrol->value.integer.value[0]);	
+	return set_mixer_volume_as_headset_gain_control_volume(
+					ucontrol->value.integer.value[0]);
 }
 
 static int __headset_playback_switch_info(struct snd_kcontrol *kcontrol,
@@ -877,7 +931,8 @@ static int __headset_playback_switch_info(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-/* When HGC_ADMUT_HED (bit 15) is 1, the headset is muted.
+/*
+ * When HGC_ADMUT_HED (bit 15) is 1, the headset is muted.
  * When HGC_ADMUT_HED is 0, headset is not muted.
  */
 static int __headset_playback_switch_get(struct snd_kcontrol *kcontrol,
@@ -891,7 +946,7 @@ static int __headset_playback_switch_get(struct snd_kcontrol *kcontrol,
 static int __headset_playback_switch_put(struct snd_kcontrol *kcontrol,
 		struct snd_ctl_elem_value *ucontrol)
 {
-	// mute/unmute headset
+	/* mute/unmute headset */
 	return adc_pga_unmute_control(ucontrol->value.integer.value[0],
 				TSC2101_HEADSET_GAIN_CTRL,
 				15);
@@ -912,22 +967,24 @@ static int __handset_playback_volume_get(struct snd_kcontrol *kcontrol,
 {
 	u16 val;
 	u16 vol;
-	
+
 	val	= omap_tsc2101_audio_read(TSC2101_HANDSET_GAIN_CTRL);
 	M_DPRINTK("registry value = %d\n", val);
 	vol	= HNGC_ADPGA_HND_EXTRACT(val);
 	vol	= vol & ~HNGC_ADMUT_HND;
 	vol	= get_handset_gain_control_volume_as_mixer_volume(vol);
 	ucontrol->value.integer.value[0]	= vol;
-	
-	M_DPRINTK("mixer volume returned = %ld\n", ucontrol->value.integer.value[0]);
+
+	M_DPRINTK("mixer volume returned = %ld\n",
+			ucontrol->value.integer.value[0]);
 	return 0;
 }
 
 static int __handset_playback_volume_put(struct snd_kcontrol *kcontrol,
 		struct snd_ctl_elem_value *ucontrol)
 {
-	return set_mixer_volume_as_handset_gain_control_volume(ucontrol->value.integer.value[0]);	
+	return set_mixer_volume_as_handset_gain_control_volume(
+					ucontrol->value.integer.value[0]);
 }
 
 static int __handset_playback_switch_info(struct snd_kcontrol *kcontrol,
@@ -940,7 +997,8 @@ static int __handset_playback_switch_info(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-/* When HNGC_ADMUT_HND (bit 15) is 1, the handset is muted.
+/*
+ * When HNGC_ADMUT_HND (bit 15) is 1, the handset is muted.
  * When HNGC_ADMUT_HND is 0, handset is not muted.
  */
 static int __handset_playback_switch_get(struct snd_kcontrol *kcontrol,
@@ -954,7 +1012,7 @@ static int __handset_playback_switch_get(struct snd_kcontrol *kcontrol,
 static int __handset_playback_switch_put(struct snd_kcontrol *kcontrol,
 		struct snd_ctl_elem_value *ucontrol)
 {
-	// handset mute/unmute
+	/* handset mute/unmute */
 	return adc_pga_unmute_control(ucontrol->value.integer.value[0],
 				TSC2101_HANDSET_GAIN_CTRL,
 				15);
@@ -970,7 +1028,8 @@ static int __cellphone_input_switch_info(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-/* When BGC_MUT_CP (bit 15) = 1, power down cellphone input pga.
+/*
+ * When BGC_MUT_CP (bit 15) = 1, power down cellphone input pga.
  * When BGC_MUT_CP = 0, power up cellphone input pga.
  */
 static int __cellphone_input_switch_get(struct snd_kcontrol *kcontrol,
@@ -986,7 +1045,7 @@ static int __cellphone_input_switch_put(struct snd_kcontrol *kcontrol,
 {
 	return adc_pga_unmute_control(ucontrol->value.integer.value[0],
 				TSC2101_BUZZER_GAIN_CTRL,
-				15);	
+				15);
 }
 
 static int __buzzer_input_switch_info(struct snd_kcontrol *kcontrol,
@@ -999,7 +1058,8 @@ static int __buzzer_input_switch_info(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-/* When BGC_MUT_BU (bit 6) = 1, power down cellphone input pga.
+/*
+ * When BGC_MUT_BU (bit 6) = 1, power down cellphone input pga.
  * When BGC_MUT_BU = 0, power up cellphone input pga.
  */
 static int __buzzer_input_switch_get(struct snd_kcontrol *kcontrol,
@@ -1015,82 +1075,82 @@ static int __buzzer_input_switch_put(struct snd_kcontrol *kcontrol,
 {
 	return adc_pga_unmute_control(ucontrol->value.integer.value[0],
 				TSC2101_BUZZER_GAIN_CTRL,
-				6);	
+				6);
 }
 
 static struct snd_kcontrol_new tsc2101_control[] __devinitdata = {
 	{
-		.name  = "Target Playback Route",
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.index = 0,
-		.access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
-		.info  = __pcm_playback_target_info,
-		.get   = __pcm_playback_target_get,
-		.put   = __pcm_playback_target_put,
+		.name	= "Target Playback Route",
+		.iface	= SNDRV_CTL_ELEM_IFACE_MIXER,
+		.index	= 0,
+		.access	= SNDRV_CTL_ELEM_ACCESS_READWRITE,
+		.info	= __pcm_playback_target_info,
+		.get	= __pcm_playback_target_get,
+		.put	= __pcm_playback_target_put,
 	}, {
-		.name  = "Master Playback Volume",
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.index = 0,
-		.access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
-		.info  = __pcm_playback_volume_info,
-		.get   = __pcm_playback_volume_get,
-		.put   = __pcm_playback_volume_put,
+		.name	= "Master Playback Volume",
+		.iface	= SNDRV_CTL_ELEM_IFACE_MIXER,
+		.index	= 0,
+		.access	= SNDRV_CTL_ELEM_ACCESS_READWRITE,
+		.info	= __pcm_playback_volume_info,
+		.get	= __pcm_playback_volume_get,
+		.put	= __pcm_playback_volume_put,
 	}, {
-		.name  = "Master Playback Switch",
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.index = 0,
-		.access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
-		.info  = __pcm_playback_switch_info,
-		.get   = __pcm_playback_switch_get,
-		.put   = __pcm_playback_switch_put,
+		.name	= "Master Playback Switch",
+		.iface	= SNDRV_CTL_ELEM_IFACE_MIXER,
+		.index	= 0,
+		.access	= SNDRV_CTL_ELEM_ACCESS_READWRITE,
+		.info	= __pcm_playback_switch_info,
+		.get	= __pcm_playback_switch_get,
+		.put	= __pcm_playback_switch_put,
 	}, {
-		.name  = "Headset Playback Volume",
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.index = 0,
-		.access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
-		.info  = __headset_playback_volume_info,
-		.get   = __headset_playback_volume_get,
-		.put   = __headset_playback_volume_put,
+		.name	= "Headset Playback Volume",
+		.iface	= SNDRV_CTL_ELEM_IFACE_MIXER,
+		.index	= 0,
+		.access	= SNDRV_CTL_ELEM_ACCESS_READWRITE,
+		.info	= __headset_playback_volume_info,
+		.get	= __headset_playback_volume_get,
+		.put	= __headset_playback_volume_put,
 	}, {
-		.name  = "Headset Playback Switch",
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.index = 0,
-		.access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
-		.info  = __headset_playback_switch_info,
-		.get   = __headset_playback_switch_get,
-		.put   = __headset_playback_switch_put,
+		.name	= "Headset Playback Switch",
+		.iface	= SNDRV_CTL_ELEM_IFACE_MIXER,
+		.index	= 0,
+		.access	= SNDRV_CTL_ELEM_ACCESS_READWRITE,
+		.info	= __headset_playback_switch_info,
+		.get	= __headset_playback_switch_get,
+		.put	= __headset_playback_switch_put,
 	}, {
-		.name  = "Handset Playback Volume",
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.index = 0,
-		.access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
-		.info  = __handset_playback_volume_info,
-		.get   = __handset_playback_volume_get,
-		.put   = __handset_playback_volume_put,
+		.name	= "Handset Playback Volume",
+		.iface	= SNDRV_CTL_ELEM_IFACE_MIXER,
+		.index	= 0,
+		.access	= SNDRV_CTL_ELEM_ACCESS_READWRITE,
+		.info	= __handset_playback_volume_info,
+		.get	= __handset_playback_volume_get,
+		.put	= __handset_playback_volume_put,
 	}, {
-		.name  = "Handset Playback Switch",
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.index = 0,
-		.access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
-		.info  = __handset_playback_switch_info,
-		.get   = __handset_playback_switch_get,
-		.put   = __handset_playback_switch_put,
+		.name	= "Handset Playback Switch",
+		.iface	= SNDRV_CTL_ELEM_IFACE_MIXER,
+		.index	= 0,
+		.access	= SNDRV_CTL_ELEM_ACCESS_READWRITE,
+		.info	= __handset_playback_switch_info,
+		.get	= __handset_playback_switch_get,
+		.put	= __handset_playback_switch_put,
 	}, {
-		.name  = "Cellphone Input Switch",
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.index = 0,
-		.access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
-		.info  = __cellphone_input_switch_info,
-		.get   = __cellphone_input_switch_get,
-		.put   = __cellphone_input_switch_put,
+		.name	= "Cellphone Input Switch",
+		.iface	= SNDRV_CTL_ELEM_IFACE_MIXER,
+		.index	= 0,
+		.access	= SNDRV_CTL_ELEM_ACCESS_READWRITE,
+		.info	= __cellphone_input_switch_info,
+		.get	= __cellphone_input_switch_get,
+		.put	= __cellphone_input_switch_put,
 	}, {
-		.name  = "Buzzer Input Switch",
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.index = 0,
-		.access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
-		.info  = __buzzer_input_switch_info,
-		.get   = __buzzer_input_switch_get,
-		.put   = __buzzer_input_switch_put,
+		.name	= "Buzzer Input Switch",
+		.iface	= SNDRV_CTL_ELEM_IFACE_MIXER,
+		.index	= 0,
+		.access	= SNDRV_CTL_ELEM_ACCESS_READWRITE,
+		.info	= __buzzer_input_switch_info,
+		.get	= __buzzer_input_switch_get,
+		.put	= __buzzer_input_switch_put,
 	}
 };
 
@@ -1106,20 +1166,20 @@ void snd_omap_resume_mixer(void)
 }
 #endif
 
-int snd_omap_mixer(struct snd_card_omap_codec *tsc2101) 
+int snd_omap_mixer(struct snd_card_omap_codec *tsc2101)
 {
-	int i=0;
-	int err=0;
+	int i = 0;
+	int err = 0;
 
-	if (!tsc2101) {
+	if (!tsc2101)
 		return -EINVAL;
-	}
-	for (i=0; i < ARRAY_SIZE(tsc2101_control); i++) {
-		if ((err = snd_ctl_add(tsc2101->card, 
-				snd_ctl_new1(&tsc2101_control[i], 
-				tsc2101->card))) < 0) {
+
+	for (i = 0; i < ARRAY_SIZE(tsc2101_control); i++) {
+		err = snd_ctl_add(tsc2101->card,
+					snd_ctl_new1(&tsc2101_control[i],
+					tsc2101->card));
+		if (err < 0)
 			return err;
-		}
 	}
 	return 0;
 }
-- 
1.5.5-rc3.GIT

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[PATCH 13/19] Code clean-up for sound/arm/omap/omap-alsa-tsc2101-mixer.h

[Eduardo Valentin]

From: Eduardo Valentin <eduardo.valentin@indt.org.br>

Removed lots of whitespaces and a few errors and
warnings reported by checkpatch.pl.

Signed-off-by: Eduardo Valentin <eduardo.valentin@indt.org.br>
---
 sound/arm/omap/omap-alsa-tsc2101-mixer.h |   29 ++++++++++++++++++-----------
 1 files changed, 18 insertions(+), 11 deletions(-)

diff --git a/sound/arm/omap/omap-alsa-tsc2101-mixer.h b/sound/arm/omap/omap-alsa-tsc2101-mixer.h
index b110b36..c0d9b23 100644
--- a/sound/arm/omap/omap-alsa-tsc2101-mixer.h
+++ b/sound/arm/omap/omap-alsa-tsc2101-mixer.h
@@ -1,9 +1,9 @@
 /*
  * sound/arm/omap/omap-alsa-tsc2101-mixer.h
- * 
+ *
  * Alsa Driver for TSC2101 codec for OMAP platform boards.
  *
- * Copyright (C) 2005 Mika Laitio <lamikr@cc.jyu.fi> and 
+ * Copyright (C) 2005 Mika Laitio <lamikr@cc.jyu.fi> and
  * 		     Everett Coleman II <gcc80x86@fuzzyneural.net>
  *
  * Based on the ideas in omap-aic23.c and sa11xx-uda1341.c
@@ -33,9 +33,9 @@
  * History:
  *
  * 2006-03-01   Mika Laitio - Mixer for the tsc2101 driver used in omap boards.
- * 		Can switch between headset and loudspeaker playback, 
+ * 		Can switch between headset and loudspeaker playback,
  * 		mute and unmute dgc, set dgc volume. Record source switch,
- * 		keyclick, buzzer and headset volume and handset volume control 
+ * 		keyclick, buzzer and headset volume and handset volume control
  * 		are still missing.
  */
 
@@ -46,8 +46,8 @@
 #include "omap-alsa-dma.h"
 
 /* tsc2101 DAC gain control volume specific  */
-#define OUTPUT_VOLUME_MIN		0x7F	// 1111111 = -63.5 DB
-#define OUTPUT_VOLUME_MAX		0x32	// 110010
+#define OUTPUT_VOLUME_MIN		0x7F	/* 1111111 = -63.5 DB */
+#define OUTPUT_VOLUME_MAX		0x32	/* 110010 */
 #define OUTPUT_VOLUME_RANGE		(OUTPUT_VOLUME_MIN - OUTPUT_VOLUME_MAX)
 
 /* use input vol of 75 for 0dB gain */
@@ -60,10 +60,15 @@
 #define PLAYBACK_TARGET_HEADPHONE	0x01
 #define PLAYBACK_TARGET_CELLPHONE	0x02
 
-/* following are used for register 03h Mixer PGA control bits D7-D5 for selecting record source */
+/*
+ * Following are used for register 03h Mixer PGA control bits D7-D5 for
+ * selecting record source
+ */
 #define REC_SRC_TARGET_COUNT		0x08
-#define REC_SRC_SINGLE_ENDED_MICIN_HED	0x00	// oss code referred to MIXER_LINE
-#define REC_SRC_SINGLE_ENDED_MICIN_HND	0x01	// oss code referred to MIXER_MIC
+/* oss code referred to MIXER_LINE */
+#define REC_SRC_SINGLE_ENDED_MICIN_HED	0x00
+/* oss code referred to MIXER_MIC */
+#define REC_SRC_SINGLE_ENDED_MICIN_HND	0x01
 #define REC_SRC_SINGLE_ENDED_AUX1	0x02
 #define REC_SRC_SINGLE_ENDED_AUX2	0x03
 #define REC_SRC_MICIN_HED_AND_AUX1	0x04
@@ -71,8 +76,10 @@
 #define REC_SRC_MICIN_HND_AND_AUX1	0x06
 #define REC_SRC_MICIN_HND_AND_AUX2	0x07
 
-#define DEFAULT_OUTPUT_VOLUME		90	// default output volume to dac dgc
-#define DEFAULT_INPUT_VOLUME		20	// default record volume
+/* default output volume to dac dgc */
+#define DEFAULT_OUTPUT_VOLUME		90
+/* default record volume */
+#define DEFAULT_INPUT_VOLUME		20
 
 #define TSC2101_AUDIO_CODEC_REGISTERS_PAGE2     (2)
 
-- 
1.5.5-rc3.GIT

[PATCH 14/19] Code clean-up for sound/arm/omap/omap-alsa-tsc2101.c

[Eduardo Valentin]

From: Eduardo Valentin <eduardo.valentin@indt.org.br>

Removed lots of whitespaces and a few errors and
warnings reported by checkpatch.pl.

Signed-off-by: Eduardo Valentin <eduardo.valentin@indt.org.br>
---
 sound/arm/omap/omap-alsa-tsc2101.c |  292 +++++++++++++++++++++++-------------
 1 files changed, 190 insertions(+), 102 deletions(-)

diff --git a/sound/arm/omap/omap-alsa-tsc2101.c b/sound/arm/omap/omap-alsa-tsc2101.c
index 1d8adc1..8a7e770 100644
--- a/sound/arm/omap/omap-alsa-tsc2101.c
+++ b/sound/arm/omap/omap-alsa-tsc2101.c
@@ -1,15 +1,15 @@
 /*
  * sound/arm/omap/omap-alsa-tsc2101.c
- * 
- * Alsa codec Driver for TSC2101 chip for OMAP platform boards. 
+ *
+ * Alsa codec Driver for TSC2101 chip for OMAP platform boards.
  * Code obtained from oss omap drivers
  *
  * Copyright (C) 2004 Texas Instruments, Inc.
  * 	Written by Nishanth Menon and Sriram Kannan
- * 	
+ *
  * Copyright (C) 2006 Instituto Nokia de Tecnologia - INdT - Manaus Brazil
  * 	Alsa modularization by Daniel Petrini (d.pensator@gmail.com)
- * 
+ *
  * Copyright (C) 2006 Mika Laitio <lamikr@cc.jyu.fi>
  *
  * This program is free software; you can redistribute it and/or modify it
@@ -23,31 +23,30 @@
 #include <linux/platform_device.h>
 #include <linux/clk.h>
 #include <linux/spi/tsc2101.h>
-#include <asm/io.h>
-#include <asm/arch/mcbsp.h>
-
+#include <linux/io.h>
 #include <linux/slab.h>
 #ifdef CONFIG_PM
 #include <linux/pm.h>
 #endif
+
 #include <asm/mach-types.h>
 #include <asm/arch/dma.h>
 #include <asm/arch/clock.h>
-
+#include <asm/arch/mcbsp.h>
 #include <asm/hardware/tsc2101.h>
-
 #include <asm/arch/omap-alsa.h>
+
 #include "omap-alsa-tsc2101.h"
 
 struct mcbsp_dev_info mcbsp_dev;
 
-static struct clk *tsc2101_mclk = 0;
+static struct clk *tsc2101_mclk;
 
-//#define DUMP_TSC2101_AUDIO_REGISTERS
+/* #define DUMP_TSC2101_AUDIO_REGISTERS */
 #undef DUMP_TSC2101_AUDIO_REGISTERS
 
 /*
- * Hardware capabilities 
+ * Hardware capabilities
  */
 
 /*
@@ -77,7 +76,7 @@ static const struct tsc2101_samplerate_reg_info
 	{8727, 6, 0},
 	/* Div 5 */
 	{8820, 5, 1},
-	{9600, 5, 0},	
+	{9600, 5, 0},
 	/* Div 4 */
 	{11025, 4, 1},
 	{12000, 4, 0},
@@ -92,22 +91,22 @@ static const struct tsc2101_samplerate_reg_info
 	{32000, 1, 0},
 	/* Div 1 */
 	{44100, 0, 1},
-	{48000, 0, 0},		
+	{48000, 0, 0},
 };
 
 static struct snd_pcm_hardware tsc2101_snd_omap_alsa_playback = {
-	.info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
-		 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID),	
+	.info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+		 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID,
 #ifdef CONFIG_MACH_OMAP_H6300
-	.formats = (SNDRV_PCM_FMTBIT_S8),
+	.formats = SNDRV_PCM_FMTBIT_S8,
 #else
- 	.formats = (SNDRV_PCM_FMTBIT_S16_LE),
+	.formats = SNDRV_PCM_FMTBIT_S16_LE,
 #endif
-	.rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |
+	.rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |
 		  SNDRV_PCM_RATE_16000 |
 		  SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |
 		  SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
-		  SNDRV_PCM_RATE_KNOT),
+		  SNDRV_PCM_RATE_KNOT,
 	.rate_min = 7350,
 	.rate_max = 48000,
 	.channels_min = 2,
@@ -121,14 +120,14 @@ static struct snd_pcm_hardware tsc2101_snd_omap_alsa_playback = {
 };
 
 static struct snd_pcm_hardware tsc2101_snd_omap_alsa_capture = {
-	.info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
-		 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID),
-	.formats = (SNDRV_PCM_FMTBIT_S16_LE),
-	.rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |
+	.info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+		 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID,
+	.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	.rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |
 		  SNDRV_PCM_RATE_16000 |
 		  SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |
 		  SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
-		  SNDRV_PCM_RATE_KNOT),
+		  SNDRV_PCM_RATE_KNOT,
 	.rate_min = 7350,
 	.rate_max = 48000,
 	.channels_min = 2,
@@ -141,7 +140,7 @@ static struct snd_pcm_hardware tsc2101_snd_omap_alsa_capture = {
 	.fifo_size = 0,
 };
 
-/* 
+/*
  * Simplified write for tsc2101 audio registers.
  */
 inline void tsc2101_audio_write(u8 address, u16 data)
@@ -150,7 +149,7 @@ inline void tsc2101_audio_write(u8 address, u16 data)
 				address, data);
 }
 
-/* 
+/*
  * Simplified read for tsc2101 audio registers.
  */
 inline u16 tsc2101_audio_read(u8 address)
@@ -160,49 +159,130 @@ inline u16 tsc2101_audio_read(u8 address)
 }
 
 #ifdef DUMP_TSC2101_AUDIO_REGISTERS
-void dump_tsc2101_audio_reg(void) {
-	printk("TSC2101_AUDIO_CTRL_1 = 0x%04x\n",	tsc2101_audio_read(TSC2101_AUDIO_CTRL_1));
-	printk("TSC2101_HEADSET_GAIN_CTRL = 0x%04x\n",	tsc2101_audio_read(TSC2101_HEADSET_GAIN_CTRL));
-	printk("TSC2101_DAC_GAIN_CTRL = 0x%04x\n", tsc2101_audio_read(TSC2101_DAC_GAIN_CTRL));
-	printk("TSC2101_MIXER_PGA_CTRL = 0x%04x\n",	tsc2101_audio_read(TSC2101_MIXER_PGA_CTRL));
-	printk("TSC2101_AUDIO_CTRL_2 = 0x%04x\n",	tsc2101_audio_read(TSC2101_AUDIO_CTRL_2));
-	printk("TSC2101_CODEC_POWER_CTRL = 0x%04x\n",	tsc2101_audio_read(TSC2101_CODEC_POWER_CTRL));
-	printk("TSC2101_AUDIO_CTRL_3 = 0x%04x\n",	tsc2101_audio_read(TSC2101_AUDIO_CTRL_3));
-	printk("TSC2101_LCH_BASS_BOOST_N0 = 0x%04x\n",	tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_N0));
-	printk("TSC2101_LCH_BASS_BOOST_N1 = 0x%04x\n",	tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_N1));
-	printk("TSC2101_LCH_BASS_BOOST_N2 = 0x%04x\n",	tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_N2));
-	printk("TSC2101_LCH_BASS_BOOST_N3 = 0x%04x\n",	tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_N3));
-	printk("TSC2101_LCH_BASS_BOOST_N4 = 0x%04x\n",	tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_N4));
-	printk("TSC2101_LCH_BASS_BOOST_N5 = 0x%04x\n",	tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_N5));
-	printk("TSC2101_LCH_BASS_BOOST_D1 = 0x%04x\n",	tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_D1));
-	printk("TSC2101_LCH_BASS_BOOST_D2 = 0x%04x\n",	tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_D2));
-	printk("TSC2101_LCH_BASS_BOOST_D4 = 0x%04x\n",	tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_D4));
-	printk("TSC2101_LCH_BASS_BOOST_D5 = 0x%04x\n",	tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_D5));
-	
-	printk("TSC2101_RCH_BASS_BOOST_N0 = 0x%04x\n",	tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_N0));
-	printk("TSC2101_RCH_BASS_BOOST_N1 = 0x%04x\n",	tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_N1));
-	printk("TSC2101_RCH_BASS_BOOST_N2 = 0x%04x\n",	tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_N2));
-	printk("TSC2101_RCH_BASS_BOOST_N3 = 0x%04x\n",	tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_N3));
-	printk("TSC2101_RCH_BASS_BOOST_N4 = 0x%04x\n",	tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_N4));
-	printk("TSC2101_RCH_BASS_BOOST_N5 = 0x%04x\n",	tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_N5));
-	printk("TSC2101_RCH_BASS_BOOST_D1 = 0x%04x\n",	tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_D1));
-	printk("TSC2101_RCH_BASS_BOOST_D2 = 0x%04x\n",	tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_D2));
-	printk("TSC2101_RCH_BASS_BOOST_D4 = 0x%04x\n",	tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_D4));
-	printk("TSC2101_RCH_BASS_BOOST_D5 = 0x%04x\n",	tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_D5));
-					
-	printk("TSC2101_PLL_PROG_1 = 0x%04x\n",	tsc2101_audio_read(TSC2101_PLL_PROG_1));
-	printk("TSC2101_PLL_PROG_1 = 0x%04x\n",	tsc2101_audio_read(TSC2101_PLL_PROG_2));
-	printk("TSC2101_AUDIO_CTRL_4 = 0x%04x\n",	tsc2101_audio_read(TSC2101_AUDIO_CTRL_4));
-	printk("TSC2101_HANDSET_GAIN_CTRL = 0x%04x\n",	tsc2101_audio_read(TSC2101_HANDSET_GAIN_CTRL));
-	printk("TSC2101_BUZZER_GAIN_CTRL = 0x%04x\n",	tsc2101_audio_read(TSC2101_BUZZER_GAIN_CTRL));
-	printk("TSC2101_AUDIO_CTRL_5 = 0x%04x\n",	tsc2101_audio_read(TSC2101_AUDIO_CTRL_5));
-	printk("TSC2101_AUDIO_CTRL_6 = 0x%04x\n",	tsc2101_audio_read(TSC2101_AUDIO_CTRL_6));
-	printk("TSC2101_AUDIO_CTRL_7 = 0x%04x\n",	tsc2101_audio_read(TSC2101_AUDIO_CTRL_7));
-	printk("TSC2101_GPIO_CTRL = 0x%04x\n",	tsc2101_audio_read(TSC2101_GPIO_CTRL));
-	printk("TSC2101_AGC_CTRL = 0x%04x\n",	tsc2101_audio_read(TSC2101_AGC_CTRL));
-	printk("TSC2101_POWERDOWN_STS = 0x%04x\n",	tsc2101_audio_read(TSC2101_POWERDOWN_STS));
-	printk("TSC2101_MIC_AGC_CONTROL = 0x%04x\n",	tsc2101_audio_read(TSC2101_MIC_AGC_CONTROL));
-	printk("TSC2101_CELL_AGC_CONTROL = 0x%04x\n",	tsc2101_audio_read(TSC2101_CELL_AGC_CONTROL));
+void dump_tsc2101_audio_reg(void)
+{
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_AUDIO_CTRL_1 = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_AUDIO_CTRL_1));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_HEADSET_GAIN_CTRL = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_HEADSET_GAIN_CTRL));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_DAC_GAIN_CTRL = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_DAC_GAIN_CTRL));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_MIXER_PGA_CTRL = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_MIXER_PGA_CTRL));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_AUDIO_CTRL_2 = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_AUDIO_CTRL_2));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_CODEC_POWER_CTRL = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_CODEC_POWER_CTRL));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_AUDIO_CTRL_3 = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_AUDIO_CTRL_3));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_LCH_BASS_BOOST_N0 = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_N0));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_LCH_BASS_BOOST_N1 = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_N1));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_LCH_BASS_BOOST_N2 = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_N2));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_LCH_BASS_BOOST_N3 = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_N3));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_LCH_BASS_BOOST_N4 = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_N4));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_LCH_BASS_BOOST_N5 = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_N5));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_LCH_BASS_BOOST_D1 = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_D1));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_LCH_BASS_BOOST_D2 = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_D2));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_LCH_BASS_BOOST_D4 = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_D4));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_LCH_BASS_BOOST_D5 = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_D5));
+
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_RCH_BASS_BOOST_N0 = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_N0));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_RCH_BASS_BOOST_N1 = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_N1));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_RCH_BASS_BOOST_N2 = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_N2));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_RCH_BASS_BOOST_N3 = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_N3));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_RCH_BASS_BOOST_N4 = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_N4));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_RCH_BASS_BOOST_N5 = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_N5));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_RCH_BASS_BOOST_D1 = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_D1));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_RCH_BASS_BOOST_D2 = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_D2));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_RCH_BASS_BOOST_D4 = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_D4));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_RCH_BASS_BOOST_D5 = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_D5));
+
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_PLL_PROG_1 = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_PLL_PROG_1));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_PLL_PROG_1 = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_PLL_PROG_2));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_AUDIO_CTRL_4 = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_AUDIO_CTRL_4));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_HANDSET_GAIN_CTRL = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_HANDSET_GAIN_CTRL));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_BUZZER_GAIN_CTRL = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_BUZZER_GAIN_CTRL));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_AUDIO_CTRL_5 = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_AUDIO_CTRL_5));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_AUDIO_CTRL_6 = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_AUDIO_CTRL_6));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_AUDIO_CTRL_7 = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_AUDIO_CTRL_7));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_GPIO_CTRL = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_GPIO_CTRL));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_AGC_CTRL = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_AGC_CTRL));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_POWERDOWN_STS = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_POWERDOWN_STS));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_MIC_AGC_CONTROL = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_MIC_AGC_CONTROL));
+	dev_dbg(&mcbsp_dev.mcbsp_dev->dev,
+		"TSC2101_CELL_AGC_CONTROL = 0x%04x\n",
+		tsc2101_audio_read(TSC2101_CELL_AGC_CONTROL));
 }
 #endif
 
@@ -233,14 +313,14 @@ void tsc2101_set_samplerate(long sample_rate)
 	if (count == NUMBER_SAMPLE_RATES_SUPPORTED) {
 		printk(KERN_ERR "Invalid Sample Rate %d requested\n",
 		       (int) sample_rate);
-		return;		// -EPERM;
+		return;
 	}
 
 	/* Set AC1 */
 	data	= tsc2101_audio_read(TSC2101_AUDIO_CTRL_1);
 	/* Clear prev settings */
 	data	&= ~(AC1_DACFS(0x07) | AC1_ADCFS(0x07));
-	data	|= AC1_DACFS(rate_reg_info[count].divisor) | 
+	data	|= AC1_DACFS(rate_reg_info[count].divisor) |
 			AC1_ADCFS(rate_reg_info[count].divisor);
 	tsc2101_audio_write(TSC2101_AUDIO_CTRL_1, data);
 
@@ -254,20 +334,25 @@ void tsc2101_set_samplerate(long sample_rate)
 #endif				/* #ifdef TSC_MASTER */
 	tsc2101_audio_write(TSC2101_AUDIO_CTRL_3, data);
 
-	/* Program the PLLs. This code assumes that the 12 Mhz MCLK is in use.
-         * If MCLK rate is something else, these values must be changed.
+	/*
+	 * Program the PLLs. This code assumes that the 12 Mhz MCLK is in use.
+	 * If MCLK rate is something else, these values must be changed.
 	 * See the tsc2101 specification for the details.
 	 */
 	if (rate_reg_info[count].fs_44kHz) {
 		/* samplerate = (44.1kHZ / x), where x is int. */
 		tsc2101_audio_write(TSC2101_PLL_PROG_1, PLL1_PLLSEL |
-				PLL1_PVAL(1) | PLL1_I_VAL(7));	/* PVAL 1; I_VAL 7 */
-		tsc2101_audio_write(TSC2101_PLL_PROG_2, PLL2_D_VAL(0x1490));	/* D_VAL 5264 */
+				/* PVAL 1; I_VAL 7 */
+				PLL1_PVAL(1) | PLL1_I_VAL(7));
+		/* D_VAL 5264 */
+		tsc2101_audio_write(TSC2101_PLL_PROG_2, PLL2_D_VAL(0x1490));
 	} else {
 		/* samplerate = (48.kHZ / x), where x is int. */
 		tsc2101_audio_write(TSC2101_PLL_PROG_1, PLL1_PLLSEL |
-			       PLL1_PVAL(1) | PLL1_I_VAL(8));	/* PVAL 1; I_VAL 8 */
-		tsc2101_audio_write(TSC2101_PLL_PROG_2, PLL2_D_VAL(0x780));	/* D_VAL 1920 */
+				/* PVAL 1; I_VAL 8 */
+			       PLL1_PVAL(1) | PLL1_I_VAL(8));
+		/* D_VAL 1920 */
+		tsc2101_audio_write(TSC2101_PLL_PROG_2, PLL2_D_VAL(0x780));
 	}
 
 	/* Set the sample rate */
@@ -295,12 +380,12 @@ void tsc2101_configure(void)
 
 /*
  *  Omap MCBSP clock and Power Management configuration
- *  
+ *
  *  Here we have some functions that allows clock to be enabled and
- *   disabled only when needed. Besides doing clock configuration 
- *   it allows turn on/turn off audio when necessary. 
+ *   disabled only when needed. Besides doing clock configuration
+ *   it allows turn on/turn off audio when necessary.
  */
- 
+
 /*
  * Do clock framework mclk search
  */
@@ -312,7 +397,7 @@ void tsc2101_clock_setup(void)
 /*
  * Do some sanity check, set clock rate, starts it and turn codec audio on
  */
-int tsc2101_clock_on(void) 
+int tsc2101_clock_on(void)
 {
 	int	curUseCount;
 	uint	curRate;
@@ -321,7 +406,7 @@ int tsc2101_clock_on(void)
 	curUseCount	= clk_get_usecount(tsc2101_mclk);
 	DPRINTK("clock use count = %d\n", curUseCount);
 	if (curUseCount > 0) {
-		// MCLK is already in use
+		/* MCLK is already in use */
 		printk(KERN_WARNING
 		       "MCLK already in use at %d Hz. We change it to %d Hz\n",
 		       (uint) clk_get_rate(tsc2101_mclk),
@@ -331,8 +416,8 @@ int tsc2101_clock_on(void)
 	if (curRate != CODEC_CLOCK) {
 		err	= clk_set_rate(tsc2101_mclk, CODEC_CLOCK);
 		if (err) {
-			printk(KERN_WARNING
-			       "Cannot set MCLK clock rate for TSC2101 CODEC, error code = %d\n", err);
+			printk(KERN_WARNING "Cannot set MCLK clock rate for "
+				"TSC2101 CODEC, error code = %d\n", err);
 			return -ECANCELED;
 		}
 	}
@@ -340,22 +425,22 @@ int tsc2101_clock_on(void)
 	curRate		= (uint)clk_get_rate(tsc2101_mclk);
 	curUseCount	= clk_get_usecount(tsc2101_mclk);
 	DPRINTK("MCLK = %d [%d], usecount = %d, clk_enable retval = %d\n",
-	       curRate, 
+	       curRate,
 	       CODEC_CLOCK,
 	       curUseCount,
 	       err);
 
-	// Now turn the audio on
+	/* Now turn the audio on */
 	tsc2101_write_sync(mcbsp_dev.tsc2101_dev, PAGE2_AUDIO_CODEC_REGISTERS,
 				TSC2101_CODEC_POWER_CTRL,
 				0x0000);
-	return 0;	
+	return 0;
 }
 
 /*
  * Do some sanity check, turn clock off and then turn codec audio off
  */
-int tsc2101_clock_off(void) 
+int tsc2101_clock_off(void)
 {
 	int curUseCount;
 	int curRate;
@@ -377,7 +462,7 @@ int tsc2101_clock_off(void)
 	tsc2101_audio_write(TSC2101_CODEC_POWER_CTRL,
 			    ~(CPC_SP1PWDN | CPC_SP2PWDN | CPC_BASSBC));
 	DPRINTK("audio codec off\n");
-	return 0;	
+	return 0;
 }
 
 int tsc2101_get_default_samplerate(void)
@@ -390,7 +475,7 @@ static int __devinit snd_omap_alsa_tsc2101_probe(struct platform_device *pdev)
 	struct spi_device *tsc2101;
 	int	ret;
 	struct	omap_alsa_codec_config *codec_cfg;
-	
+
 	tsc2101 = dev_get_drvdata(&pdev->dev);
 	if (tsc2101 == NULL) {
 		dev_err(&pdev->dev, "no platform data\n");
@@ -405,18 +490,21 @@ static int __devinit snd_omap_alsa_tsc2101_probe(struct platform_device *pdev)
 
 	codec_cfg = pdev->dev.platform_data;
 	if (codec_cfg != NULL) {
-		codec_cfg->hw_constraints_rates	= &tsc2101_hw_constraints_rates;
-		codec_cfg->snd_omap_alsa_playback  = &tsc2101_snd_omap_alsa_playback;
-		codec_cfg->snd_omap_alsa_capture  = &tsc2101_snd_omap_alsa_capture;
+		codec_cfg->hw_constraints_rates	=
+						&tsc2101_hw_constraints_rates;
+		codec_cfg->snd_omap_alsa_playback =
+					&tsc2101_snd_omap_alsa_playback;
+		codec_cfg->snd_omap_alsa_capture =
+						&tsc2101_snd_omap_alsa_capture;
 		codec_cfg->codec_configure_dev	= tsc2101_configure;
 		codec_cfg->codec_set_samplerate	= tsc2101_set_samplerate;
 		codec_cfg->codec_clock_setup	= tsc2101_clock_setup;
 		codec_cfg->codec_clock_on	= tsc2101_clock_on;
 		codec_cfg->codec_clock_off	= tsc2101_clock_off;
-		codec_cfg->get_default_samplerate = tsc2101_get_default_samplerate;
-		ret	= snd_omap_alsa_post_probe(pdev, codec_cfg);
-	}
-	else
+		codec_cfg->get_default_samplerate =
+						tsc2101_get_default_samplerate;
+		ret = snd_omap_alsa_post_probe(pdev, codec_cfg);
+	} else
 		ret = -ENODEV;
 	return ret;
 }
@@ -432,10 +520,10 @@ static struct platform_driver omap_alsa_driver = {
 };
 
 static int __init omap_alsa_tsc2101_init(void)
-{	
+{
 	ADEBUG();
 #ifdef DUMP_TSC2101_AUDIO_REGISTERS
-	printk("omap_alsa_tsc2101_init()\n");
+	printk(KERN_INFO "omap_alsa_tsc2101_init()\n");
 	dump_tsc2101_audio_reg();
 #endif
 	return platform_driver_register(&omap_alsa_driver);
@@ -445,7 +533,7 @@ static void __exit omap_alsa_tsc2101_exit(void)
 {
 	ADEBUG();
 #ifdef DUMP_TSC2101_AUDIO_REGISTERS
-	printk("omap_alsa_tsc2101_exit()\n");
+	printk(KERN_INFO "omap_alsa_tsc2101_exit()\n");
 	dump_tsc2101_audio_reg();
 #endif
 	platform_driver_unregister(&omap_alsa_driver);
-- 
1.5.5-rc3.GIT

[PATCH 15/19] Code clean-up for sound/arm/omap/omap-alsa-tsc2101.h

[Eduardo Valentin]

From: Eduardo Valentin <eduardo.valentin@indt.org.br>

Removed lots of whitespaces and a few errors and
warnings reported by checkpatch.pl.

Signed-off-by: Eduardo Valentin <eduardo.valentin@indt.org.br>
---
 sound/arm/omap/omap-alsa-tsc2101.h |    6 +++---
 1 files changed, 3 insertions(+), 3 deletions(-)

diff --git a/sound/arm/omap/omap-alsa-tsc2101.h b/sound/arm/omap/omap-alsa-tsc2101.h
index 73999a0..fb03801 100644
--- a/sound/arm/omap/omap-alsa-tsc2101.h
+++ b/sound/arm/omap/omap-alsa-tsc2101.h
@@ -1,6 +1,6 @@
 /*
  * sound/arm/omap/omap-alsa-tsc2101.h
- * 
+ *
  * Alsa Driver for TSC2101 codec for OMAP platform boards.
  *
  * Based on former omap-aic23.h and tsc2101 OSS drivers.
@@ -9,7 +9,7 @@
  *
  * Copyright (C) 2006 Instituto Nokia de Tecnologia - INdT - Manaus Brazil
  *          Alsa modularization by Daniel Petrini (d.pensator@gmail.com)
- * 
+ *
  * Copyright (C) 2006 Mika Laitio <lamikr@cc.jyu.fi>
  *
  * This program is free software; you can redistribute it and/or modify it
@@ -54,7 +54,7 @@ struct tsc2101_samplerate_reg_info {
 };
 
 /*
- * Defines codec specific functions pointers that can be used from the 
+ * Defines codec specific function pointers that can be used from the
  * common omap-alse base driver for all omap codecs. (tsc2101 and aic23)
  */
 inline void tsc2101_configure(void);
-- 
1.5.5-rc3.GIT

[PATCH 16/19] Code clean-up for sound/arm/omap/omap-alsa-tsc2102-mixer.c

[Eduardo Valentin]

From: Eduardo Valentin <eduardo.valentin@indt.org.br>

Removed lots of whitespaces and a few errors and
warnings reported by checkpatch.pl.

Signed-off-by: Eduardo Valentin <eduardo.valentin@indt.org.br>
---
 sound/arm/omap/omap-alsa-tsc2102-mixer.c |   14 ++++++++------
 1 files changed, 8 insertions(+), 6 deletions(-)

diff --git a/sound/arm/omap/omap-alsa-tsc2102-mixer.c b/sound/arm/omap/omap-alsa-tsc2102-mixer.c
index 99b1a1d..cf1f852 100644
--- a/sound/arm/omap/omap-alsa-tsc2102-mixer.c
+++ b/sound/arm/omap/omap-alsa-tsc2102-mixer.c
@@ -82,8 +82,10 @@ void init_playback_targets(void)
 	/* Unmute */
 	tsc2102_set_mute(0, 0);
 
-	mute[0] = mute[1] = 0;
-	filter[0] = filter[1] = 0;
+	mute[0] = 0;
+	mute[1] = 0;
+	filter[0] = 0;
+	filter[1] = 0;
 }
 
 /*
@@ -146,7 +148,7 @@ static int __pcm_playback_switch_get(struct snd_kcontrol *kcontrol,
 }
 
 static int __pcm_playback_switch_put(struct snd_kcontrol *kcontrol,
-		struct snd_ctl_elem_value *ucontrol) 
+		struct snd_ctl_elem_value *ucontrol)
 {
 	mute[0] = (ucontrol->value.integer.value[0] == 0);	/* L */
 	mute[1] = (ucontrol->value.integer.value[1] == 0);	/* R */
@@ -173,7 +175,7 @@ static int __pcm_playback_deemphasis_get(struct snd_kcontrol *kcontrol,
 }
 
 static int __pcm_playback_deemphasis_put(struct snd_kcontrol *kcontrol,
-		struct snd_ctl_elem_value *ucontrol) 
+		struct snd_ctl_elem_value *ucontrol)
 {
 	filter[0] = (ucontrol->value.integer.value[0] > 0);
 
@@ -199,7 +201,7 @@ static int __pcm_playback_bassboost_get(struct snd_kcontrol *kcontrol,
 }
 
 static int __pcm_playback_bassboost_put(struct snd_kcontrol *kcontrol,
-		struct snd_ctl_elem_value *ucontrol) 
+		struct snd_ctl_elem_value *ucontrol)
 {
 	filter[1] = (ucontrol->value.integer.value[0] > 0);
 
@@ -270,7 +272,7 @@ int snd_omap_mixer(struct snd_card_omap_codec *tsc2102)
 	if (!tsc2102)
 		return -EINVAL;
 
-	for (i = 0; i < ARRAY_SIZE(tsc2102_controls); i ++) {
+	for (i = 0; i < ARRAY_SIZE(tsc2102_controls); i++) {
 		err = snd_ctl_add(tsc2102->card,
 				snd_ctl_new1(&tsc2102_controls[i],
 				tsc2102->card));
-- 
1.5.5-rc3.GIT

[PATCH 17/19] Code clean-up for sound/arm/omap/omap-alsa-tsc2102.c

[Eduardo Valentin]

From: Eduardo Valentin <eduardo.valentin@indt.org.br>

Removed lots of whitespaces and a few errors and
warnings reported by checkpatch.pl.

Signed-off-by: Eduardo Valentin <eduardo.valentin@indt.org.br>
---
 sound/arm/omap/omap-alsa-tsc2102.c |   70 ++++++++++++++++++------------------
 1 files changed, 35 insertions(+), 35 deletions(-)

diff --git a/sound/arm/omap/omap-alsa-tsc2102.c b/sound/arm/omap/omap-alsa-tsc2102.c
index 1ca5ba9..911e776 100644
--- a/sound/arm/omap/omap-alsa-tsc2102.c
+++ b/sound/arm/omap/omap-alsa-tsc2102.c
@@ -1,6 +1,6 @@
 /*
  * sound/arm/omap/omap-alsa-tsc2102.c
- * 
+ *
  * Alsa codec driver for TSC2102 chip for OMAP platforms.
  *
  * Copyright (c) 2006 Andrzej Zaborowski  <balrog@zabor.org>
@@ -17,17 +17,16 @@
 #include <linux/platform_device.h>
 #include <linux/clk.h>
 #include <linux/module.h>
+#include <linux/io.h>
 #include <linux/spi/tsc2102.h>
 
-#include <asm/io.h>
-
 #include <asm/arch/dma.h>
 #include <asm/arch/clock.h>
 #include <asm/arch/omap-alsa.h>
 
 #include "omap-alsa-tsc2102.h"
 
-static struct clk *tsc2102_bclk = 0;
+static struct clk *tsc2102_bclk;
 
 /*
  * Hardware capabilities
@@ -67,62 +66,63 @@ static struct snd_pcm_hardware tsc2102_snd_omap_alsa_playback = {
 };
 
 #ifdef DUMP_TSC2102_AUDIO_REGISTERS
-static void dump_tsc2102_audio_regs(void) {
-	printk("TSC2102_AUDIO1_CTRL = 0x%04x\n",
+static void dump_tsc2102_audio_regs(void)
+{
+	printk(KERN_INFO "TSC2102_AUDIO1_CTRL = 0x%04x\n",
 			tsc2102_read_sync(TSC2102_AUDIO1_CTRL));
-	printk("TSC2102_DAC_GAIN_CTRL = 0x%04x\n",
+	printk(KERN_INFO "TSC2102_DAC_GAIN_CTRL = 0x%04x\n",
 			tsc2102_read_sync(TSC2102_DAC_GAIN_CTRL));
-	printk("TSC2102_AUDIO2_CTRL = 0x%04x\n",
+	printk(KERN_INFO "TSC2102_AUDIO2_CTRL = 0x%04x\n",
 			tsc2102_read_sync(TSC2102_AUDIO2_CTRL));
-	printk("TSC2102_DAC_POWER_CTRL = 0x%04x\n",
+	printk(KERN_INFO "TSC2102_DAC_POWER_CTRL = 0x%04x\n",
 			tsc2102_read_sync(TSC2102_DAC_POWER_CTRL));
-	printk("TSC2102_AUDIO3_CTRL = 0x%04x\n",
+	printk(KERN_INFO "TSC2102_AUDIO3_CTRL = 0x%04x\n",
 			tsc2102_read_sync(TSC2102_AUDIO_CTRL_3));
-	printk("TSC2102_LCH_BASS_BOOST_N0 = 0x%04x\n",
+	printk(KERN_INFO "TSC2102_LCH_BASS_BOOST_N0 = 0x%04x\n",
 			tsc2102_read_sync(TSC2102_LCH_BASS_BOOST_N0));
-	printk("TSC2102_LCH_BASS_BOOST_N1 = 0x%04x\n",
+	printk(KERN_INFO "TSC2102_LCH_BASS_BOOST_N1 = 0x%04x\n",
 			tsc2102_read_sync(TSC2102_LCH_BASS_BOOST_N1));
-	printk("TSC2102_LCH_BASS_BOOST_N2 = 0x%04x\n",
+	printk(KERN_INFO "TSC2102_LCH_BASS_BOOST_N2 = 0x%04x\n",
 			tsc2102_read_sync(TSC2102_LCH_BASS_BOOST_N2));
-	printk("TSC2102_LCH_BASS_BOOST_N3 = 0x%04x\n",
+	printk(KERN_INFO "TSC2102_LCH_BASS_BOOST_N3 = 0x%04x\n",
 			tsc2102_read_sync(TSC2102_LCH_BASS_BOOST_N3));
-	printk("TSC2102_LCH_BASS_BOOST_N4 = 0x%04x\n",
+	printk(KERN_INFO "TSC2102_LCH_BASS_BOOST_N4 = 0x%04x\n",
 			tsc2102_read_sync(TSC2102_LCH_BASS_BOOST_N4));
-	printk("TSC2102_LCH_BASS_BOOST_N5 = 0x%04x\n",
+	printk(KERN_INFO "TSC2102_LCH_BASS_BOOST_N5 = 0x%04x\n",
 			tsc2102_read_sync(TSC2102_LCH_BASS_BOOST_N5));
-	printk("TSC2102_LCH_BASS_BOOST_D1 = 0x%04x\n",
+	printk(KERN_INFO "TSC2102_LCH_BASS_BOOST_D1 = 0x%04x\n",
 			tsc2102_read_sync(TSC2102_LCH_BASS_BOOST_D1));
-	printk("TSC2102_LCH_BASS_BOOST_D2 = 0x%04x\n",
+	printk(KERN_INFO "TSC2102_LCH_BASS_BOOST_D2 = 0x%04x\n",
 			tsc2102_read_sync(TSC2102_LCH_BASS_BOOST_D2));
-	printk("TSC2102_LCH_BASS_BOOST_D4 = 0x%04x\n",
+	printk(KERN_INFO "TSC2102_LCH_BASS_BOOST_D4 = 0x%04x\n",
 			tsc2102_read_sync(TSC2102_LCH_BASS_BOOST_D4));
-	printk("TSC2102_LCH_BASS_BOOST_D5 = 0x%04x\n",
+	printk(KERN_INFO "TSC2102_LCH_BASS_BOOST_D5 = 0x%04x\n",
 			tsc2102_read_sync(TSC2102_LCH_BASS_BOOST_D5));
-	printk("TSC2102_RCH_BASS_BOOST_N0 = 0x%04x\n",
+	printk(KERN_INFO "TSC2102_RCH_BASS_BOOST_N0 = 0x%04x\n",
 			tsc2102_read_sync(TSC2102_RCH_BASS_BOOST_N0));
-	printk("TSC2102_RCH_BASS_BOOST_N1 = 0x%04x\n",
+	printk(KERN_INFO "TSC2102_RCH_BASS_BOOST_N1 = 0x%04x\n",
 			tsc2102_read_sync(TSC2102_RCH_BASS_BOOST_N1));
-	printk("TSC2102_RCH_BASS_BOOST_N2 = 0x%04x\n",
+	printk(KERN_INFO "TSC2102_RCH_BASS_BOOST_N2 = 0x%04x\n",
 			tsc2102_read_sync(TSC2102_RCH_BASS_BOOST_N2));
-	printk("TSC2102_RCH_BASS_BOOST_N3 = 0x%04x\n",
+	printk(KERN_INFO "TSC2102_RCH_BASS_BOOST_N3 = 0x%04x\n",
 			tsc2102_read_sync(TSC2102_RCH_BASS_BOOST_N3));
-	printk("TSC2102_RCH_BASS_BOOST_N4 = 0x%04x\n",
+	printk(KERN_INFO "TSC2102_RCH_BASS_BOOST_N4 = 0x%04x\n",
 			tsc2102_read_sync(TSC2102_RCH_BASS_BOOST_N4));
-	printk("TSC2102_RCH_BASS_BOOST_N5 = 0x%04x\n",
+	printk(KERN_INFO "TSC2102_RCH_BASS_BOOST_N5 = 0x%04x\n",
 			tsc2102_read_sync(TSC2102_RCH_BASS_BOOST_N5));
-	printk("TSC2102_RCH_BASS_BOOST_D1 = 0x%04x\n",
+	printk(KERN_INFO "TSC2102_RCH_BASS_BOOST_D1 = 0x%04x\n",
 			tsc2102_read_sync(TSC2102_RCH_BASS_BOOST_D1));
-	printk("TSC2102_RCH_BASS_BOOST_D2 = 0x%04x\n",
+	printk(KERN_INFO "TSC2102_RCH_BASS_BOOST_D2 = 0x%04x\n",
 			tsc2102_read_sync(TSC2102_RCH_BASS_BOOST_D2));
-	printk("TSC2102_RCH_BASS_BOOST_D4 = 0x%04x\n",
+	printk(KERN_INFO "TSC2102_RCH_BASS_BOOST_D4 = 0x%04x\n",
 			tsc2102_read_sync(TSC2102_RCH_BASS_BOOST_D4));
-	printk("TSC2102_RCH_BASS_BOOST_D5 = 0x%04x\n",
+	printk(KERN_INFO "TSC2102_RCH_BASS_BOOST_D5 = 0x%04x\n",
 			tsc2102_read_sync(TSC2102_RCH_BASS_BOOST_D5));
-	printk("TSC2102_PLL1_CTRL = 0x%04x\n",
+	printk(KERN_INFO "TSC2102_PLL1_CTRL = 0x%04x\n",
 			tsc2102_read_sync(TSC2102_PLL1_CTRL));
-	printk("TSC2102_PLL2_CTRL = 0x%04x\n",
+	printk(KERN_INFO "TSC2102_PLL2_CTRL = 0x%04x\n",
 			tsc2102_read_sync(TSC2102_PLL2_CTRL));
-	printk("TSC2102_AUDIO4_CTRL = 0x%04x\n",
+	printk(KERN_INFO "TSC2102_AUDIO4_CTRL = 0x%04x\n",
 			tsc2102_read_sync(TSC2102_AUDIO4_CTRL));
 }
 #endif
@@ -131,7 +131,7 @@ static void dump_tsc2102_audio_regs(void) {
  * ALSA operations according to board file
  */
 
-static long current_rate = 0;
+static long current_rate;
 
 /*
  * Sample rate changing
@@ -180,7 +180,7 @@ static void tsc2102_configure(void)
 
 /*
  * Omap McBSP clock and Power Management configuration
- *  
+ *
  * Here we have some functions that allow clock to be enabled and
  * disabled only when needed.  Besides doing clock configuration
  * they allow turn audio on and off when necessary.
-- 
1.5.5-rc3.GIT

[PATCH 18/19] Code clean-up for sound/arm/omap/omap-alsa-tsc2102.h

[Eduardo Valentin]

From: Eduardo Valentin <eduardo.valentin@indt.org.br>

Removed lots of whitespaces and a few errors and
warnings reported by checkpatch.pl.

Signed-off-by: Eduardo Valentin <eduardo.valentin@indt.org.br>
---
 sound/arm/omap/omap-alsa-tsc2102.h |    2 +-
 1 files changed, 1 insertions(+), 1 deletions(-)

diff --git a/sound/arm/omap/omap-alsa-tsc2102.h b/sound/arm/omap/omap-alsa-tsc2102.h
index cee2839..0e0c097 100644
--- a/sound/arm/omap/omap-alsa-tsc2102.h
+++ b/sound/arm/omap/omap-alsa-tsc2102.h
@@ -1,6 +1,6 @@
 /*
  * sound/arm/omap/omap-alsa-tsc2102.h
- * 
+ *
  * Alsa codec driver for TSC2102 chip for OMAP platforms.
  *
  * Copyright (c) 2006 Andrzej Zaborowski  <balrog@zabor.org>
-- 
1.5.5-rc3.GIT

[PATCH 19/19] Code clean-up for sound/arm/omap/omap-alsa.c

[Eduardo Valentin]

From: Eduardo Valentin <eduardo.valentin@indt.org.br>

Removed lots of whitespaces and a few errors and
warnings reported by checkpatch.pl.

Signed-off-by: Eduardo Valentin <eduardo.valentin@indt.org.br>
---
 sound/arm/omap/omap-alsa.c |  163 +++++++++++++++++++++++---------------------
 1 files changed, 86 insertions(+), 77 deletions(-)

diff --git a/sound/arm/omap/omap-alsa.c b/sound/arm/omap/omap-alsa.c
index 30c7942..8693942 100644
--- a/sound/arm/omap/omap-alsa.c
+++ b/sound/arm/omap/omap-alsa.c
@@ -1,6 +1,6 @@
 /*
  * sound/arm/omap-alsa.c
- * 
+ *
  * Alsa Driver for OMAP
  *
  * Copyright (C) 2005 Instituto Nokia de Tecnologia - INdT - Manaus Brazil
@@ -9,7 +9,7 @@
  *
  * Copyright (C) 2006 Mika Laitio <lamikr@cc.jyu.fi>
  *
- * Based on sa11xx-uda1341.c, 
+ * Based on sa11xx-uda1341.c,
  * Copyright (C) 2002 Tomas Kasparek <tomas.kasparek@seznam.cz>
  *
  * This program is free software; you can redistribute it and/or modify it
@@ -34,10 +34,10 @@
  *
  * History:
  *
- * 2005-07-29   INdT Kernel Team - Alsa driver for omap osk. Creation of new 
+ * 2005-07-29   INdT Kernel Team - Alsa driver for omap osk. Creation of new
  *                                 file omap-aic23.c
- * 
- * 2005-12-18   Dirk Behme       - Added L/R Channel Interchange fix as proposed 
+ *
+ * 2005-12-18   Dirk Behme       - Added L/R Channel Interchange fix as proposed
  *                                 by Ajaya Babu
  *
  */
@@ -52,14 +52,18 @@
 #include <asm/arch/omap-alsa.h>
 #include "omap-alsa-dma.h"
 
-MODULE_AUTHOR("Mika Laitio, Daniel Petrini, David Cohen, Anderson Briglia - INdT");
+MODULE_AUTHOR("Mika Laitio");
+MODULE_AUTHOR("Daniel Petrini");
+MODULE_AUTHOR("David Cohen");
+MODULE_AUTHOR("Anderson Briglia");
+
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("OMAP driver for ALSA");
 MODULE_ALIAS("omap_alsa_mcbsp.1");
 
-static char *id	= NULL;	
-static struct snd_card_omap_codec 	*alsa_codec		= NULL;
-static struct omap_alsa_codec_config	*alsa_codec_config	= NULL;
+static char *id;
+static struct snd_card_omap_codec 	*alsa_codec;
+static struct omap_alsa_codec_config	*alsa_codec_config;
 
 /*
  * HW interface start and stop helper functions
@@ -100,10 +104,9 @@ static void omap_alsa_audio_init(struct snd_card_omap_codec *omap_alsa)
 	    audio_ifc_stop;
 }
 
-/* 
- * DMA functions 
+/*
+ * DMA functions
  * Depends on omap-alsa-dma.c functions and (omap) dma.c
- * 
  */
 static int audio_dma_request(struct audio_stream *s,
 			     void (*callback) (void *))
@@ -151,7 +154,8 @@ static u_int audio_get_dma_pos(struct audio_stream *s)
 	spin_unlock_irqrestore(&s->dma_lock, flags);
 
 	/* Now, the position related to the end of that period */
-	offset = bytes_to_frames(runtime, s->offset) - bytes_to_frames(runtime, count);
+	offset = bytes_to_frames(runtime, s->offset) -
+			bytes_to_frames(runtime, count);
 
 	if (offset >= runtime->buffer_size)
 		offset = 0;
@@ -190,29 +194,28 @@ static void audio_process_dma(struct audio_stream *s)
 	unsigned int dma_size;
 	unsigned int offset;
 	int ret;
-	
+
 	ADEBUG();
 	runtime = substream->runtime;
 	if (s->active) {
 		dma_size = frames_to_bytes(runtime, runtime->period_size);
 		offset = dma_size * s->period;
-		snd_assert(dma_size <= DMA_BUF_SIZE,);
+		snd_assert(dma_size <= DMA_BUF_SIZE, return);
 		/*
 		 * On omap1510 based devices, we need to call the stop_dma
 		 * before calling the start_dma or we will not receive the
 		 * irq from DMA after the first transfered/played buffer.
 		 * (invocation of callback_omap_alsa_sound_dma() method).
 		 */
-		if (cpu_is_omap1510()) {
+		if (cpu_is_omap1510())
 			omap_stop_alsa_sound_dma(s);
-		}
+
 		ret = omap_start_alsa_sound_dma(s,
 				(dma_addr_t)runtime->dma_area + offset,
 				dma_size);
 		if (ret) {
-			printk(KERN_ERR
-			       "audio_process_dma: cannot queue DMA buffer (%i)\n",
-			       ret);
+			printk(KERN_ERR "audio_process_dma: cannot"
+					" queue DMA buffer (%i)\n", ret);
 			return;
 		}
 
@@ -223,15 +226,15 @@ static void audio_process_dma(struct audio_stream *s)
 	}
 }
 
-/* 
+/*
  *  This is called when dma IRQ occurs at the end of each transmited block
  */
 void callback_omap_alsa_sound_dma(void *data)
 {
 	struct audio_stream *s = data;
-	
+
 	ADEBUG();
-	/* 
+	/*
 	 * If we are getting a callback for an active stream then we inform
 	 * the PCM middle layer we've finished a period
 	 */
@@ -239,25 +242,25 @@ void callback_omap_alsa_sound_dma(void *data)
 		snd_pcm_period_elapsed(s->stream);
 
 	spin_lock(&s->dma_lock);
-	if (s->periods > 0) 
+	if (s->periods > 0)
 		s->periods--;
-	
+
 	audio_process_dma(s);
 	spin_unlock(&s->dma_lock);
 }
 
-/* 
+/*
  * Alsa section
  * PCM settings and callbacks
  */
-static int snd_omap_alsa_trigger(struct snd_pcm_substream * substream, int cmd)
+static int snd_omap_alsa_trigger(struct snd_pcm_substream *substream, int cmd)
 {
 	struct snd_card_omap_codec *chip =
 	    snd_pcm_substream_chip(substream);
 	int stream_id = substream->pstr->stream;
 	struct audio_stream *s = &chip->s[stream_id];
 	int err = 0;
-	
+
 	ADEBUG();
 	/* note local interrupts are already disabled in the midlevel code */
 	spin_lock(&s->dma_lock);
@@ -276,16 +279,16 @@ static int snd_omap_alsa_trigger(struct snd_pcm_substream * substream, int cmd)
 		break;
 	}
 	spin_unlock(&s->dma_lock);
-	
+
 	return err;
 }
 
-static int snd_omap_alsa_prepare(struct snd_pcm_substream * substream)
+static int snd_omap_alsa_prepare(struct snd_pcm_substream *substream)
 {
 	struct snd_card_omap_codec *chip = snd_pcm_substream_chip(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct audio_stream *s = &chip->s[substream->pstr->stream];
-	
+
 	ADEBUG();
 	/* set requested samplerate */
 	alsa_codec_config->codec_set_samplerate(runtime->rate);
@@ -297,63 +300,64 @@ static int snd_omap_alsa_prepare(struct snd_pcm_substream * substream)
 	return 0;
 }
 
-static snd_pcm_uframes_t snd_omap_alsa_pointer(struct snd_pcm_substream *substream)
+static snd_pcm_uframes_t
+snd_omap_alsa_pointer(struct snd_pcm_substream *substream)
 {
 	struct snd_card_omap_codec *chip = snd_pcm_substream_chip(substream);
 
-	ADEBUG();	
+	ADEBUG();
 	return audio_get_dma_pos(&chip->s[substream->pstr->stream]);
 }
 
-static int snd_card_omap_alsa_open(struct snd_pcm_substream * substream)
+static int snd_card_omap_alsa_open(struct snd_pcm_substream *substream)
 {
 	struct snd_card_omap_codec *chip =
 	    snd_pcm_substream_chip(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	int stream_id = substream->pstr->stream;
 	int err;
-	
+
 	ADEBUG();
 	chip->s[stream_id].stream = substream;
 	alsa_codec_config->codec_clock_on();
-	if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) 
+	if (stream_id == SNDRV_PCM_STREAM_PLAYBACK)
 		runtime->hw = *(alsa_codec_config->snd_omap_alsa_playback);
-	else 
+	else
 		runtime->hw = *(alsa_codec_config->snd_omap_alsa_capture);
-	
-	if ((err = snd_pcm_hw_constraint_integer(runtime,
-					   SNDRV_PCM_HW_PARAM_PERIODS)) < 0) 
+
+	err = snd_pcm_hw_constraint_integer(runtime,
+						SNDRV_PCM_HW_PARAM_PERIODS);
+	if (err < 0)
 		return err;
-	
-	if ((err = snd_pcm_hw_constraint_list(runtime,
-					0,
-					SNDRV_PCM_HW_PARAM_RATE,
-					alsa_codec_config->hw_constraints_rates)) < 0) 
+
+	err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+				alsa_codec_config->hw_constraints_rates);
+	if (err < 0)
 		return err;
-	
+
 	return 0;
 }
 
-static int snd_card_omap_alsa_close(struct snd_pcm_substream * substream)
+static int snd_card_omap_alsa_close(struct snd_pcm_substream *substream)
 {
 	struct snd_card_omap_codec *chip = snd_pcm_substream_chip(substream);
-	
+
 	ADEBUG();
 	alsa_codec_config->codec_clock_off();
 	chip->s[substream->pstr->stream].stream = NULL;
-	
+
 	return 0;
 }
 
 /* HW params & free */
-static int snd_omap_alsa_hw_params(struct snd_pcm_substream * substream,
-				   struct snd_pcm_hw_params * hw_params)
+static int snd_omap_alsa_hw_params(struct snd_pcm_substream *substream,
+					struct snd_pcm_hw_params *hw_params)
 {
 	return snd_pcm_lib_malloc_pages(substream,
 					params_buffer_bytes(hw_params));
 }
 
-static int snd_omap_alsa_hw_free(struct snd_pcm_substream * substream)
+static int snd_omap_alsa_hw_free(struct snd_pcm_substream *substream)
 {
 	return snd_pcm_lib_free_pages(substream);
 }
@@ -383,17 +387,17 @@ static struct snd_pcm_ops snd_card_omap_alsa_capture_ops = {
 
 /*
  *  Alsa init and exit section
- *  
  *  Inits pcm alsa structures, allocate the alsa buffer, suspend, resume
  */
-static int __init snd_card_omap_alsa_pcm(struct snd_card_omap_codec *omap_alsa, 
+static int __init snd_card_omap_alsa_pcm(struct snd_card_omap_codec *omap_alsa,
 					int device)
 {
 	struct snd_pcm *pcm;
 	int err;
-	
+
 	ADEBUG();
-	if ((err = snd_pcm_new(omap_alsa->card, "OMAP PCM", device, 1, 1, &pcm)) < 0)
+	err = snd_pcm_new(omap_alsa->card, "OMAP PCM", device, 1, 1, &pcm);
+	if (err < 0)
 		return err;
 
 	/* sets up initial buffer with continuous allocation */
@@ -433,11 +437,12 @@ int snd_omap_alsa_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_card_omap_codec *chip;
 	struct snd_card *card = platform_get_drvdata(pdev);
-	
+
 	if (card->power_state != SNDRV_CTL_POWER_D3hot) {
 		chip = card->private_data;
 		if (chip->card->power_state != SNDRV_CTL_POWER_D3hot) {
-			snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot);
+			snd_power_change_state(chip->card,
+						SNDRV_CTL_POWER_D3hot);
 			snd_pcm_suspend_all(chip->pcm);
 			/* Mutes and turn clock off */
 			alsa_codec_config->codec_clock_off();
@@ -452,7 +457,7 @@ int snd_omap_alsa_resume(struct platform_device *pdev)
 	struct snd_card_omap_codec *chip;
 	struct snd_card *card = platform_get_drvdata(pdev);
 
-	if (card->power_state != SNDRV_CTL_POWER_D0) {				
+	if (card->power_state != SNDRV_CTL_POWER_D0) {
 		chip = card->private_data;
 		if (chip->card->power_state != SNDRV_CTL_POWER_D0) {
 			snd_power_change_state(chip->card, SNDRV_CTL_POWER_D0);
@@ -465,11 +470,11 @@ int snd_omap_alsa_resume(struct platform_device *pdev)
 
 #endif	/* CONFIG_PM */
 
-void snd_omap_alsa_free(struct snd_card * card)
+void snd_omap_alsa_free(struct snd_card *card)
 {
 	struct snd_card_omap_codec *chip = card->private_data;
 	ADEBUG();
-	
+
 	/*
 	 * Turn off codec after it is done.
 	 * Can't do it immediately, since it may still have
@@ -486,27 +491,28 @@ void snd_omap_alsa_free(struct snd_card * card)
 
 /* module init & exit */
 
-/* 
+/*
  * Inits alsa soudcard structure.
  * Called by the probe method in codec after function pointers has been set.
  */
-int snd_omap_alsa_post_probe(struct platform_device *pdev, struct omap_alsa_codec_config *config)
+int snd_omap_alsa_post_probe(struct platform_device *pdev,
+				struct omap_alsa_codec_config *config)
 {
 	int err = 0;
 	int def_rate;
 	struct snd_card *card;
-	
+
 	ADEBUG();
 	alsa_codec_config	= config;
 
 	alsa_codec_config->codec_clock_setup();
-	alsa_codec_config->codec_clock_on(); 
+	alsa_codec_config->codec_clock_on();
 
 	omap_mcbsp_request(AUDIO_MCBSP);
 	omap_mcbsp_stop(AUDIO_MCBSP);
 	omap_mcbsp_config(AUDIO_MCBSP, alsa_codec_config->mcbsp_regs_alsa);
 	omap_mcbsp_start(AUDIO_MCBSP);
-	
+
 	if (alsa_codec_config && alsa_codec_config->codec_configure_dev)
 		alsa_codec_config->codec_configure_dev();
 
@@ -525,18 +531,20 @@ int snd_omap_alsa_post_probe(struct platform_device *pdev, struct omap_alsa_code
 	card->private_free = snd_omap_alsa_free;
 
 	alsa_codec->card	= card;
-	def_rate		= alsa_codec_config->get_default_samplerate(); 
+	def_rate		= alsa_codec_config->get_default_samplerate();
 	alsa_codec->samplerate	= def_rate;
 
 	spin_lock_init(&alsa_codec->s[0].dma_lock);
 	spin_lock_init(&alsa_codec->s[1].dma_lock);
 
 	/* mixer */
-	if ((err = snd_omap_mixer(alsa_codec)) < 0)
+	err = snd_omap_mixer(alsa_codec);
+	if (err < 0)
 		goto nodev3;
 
 	/* PCM */
-	if ((err = snd_card_omap_alsa_pcm(alsa_codec, 0)) < 0)
+	err = snd_card_omap_alsa_pcm(alsa_codec, 0);
+	if (err < 0)
 		goto nodev3;
 
 	strcpy(card->driver, "OMAP_ALSA");
@@ -545,16 +553,17 @@ int snd_omap_alsa_post_probe(struct platform_device *pdev, struct omap_alsa_code
 
 	snd_omap_init_mixer();
 	snd_card_set_dev(card, &pdev->dev);
-	
-	if ((err = snd_card_register(card)) == 0) {
+
+	err = snd_card_register(card);
+	if (err == 0) {
 		printk(KERN_INFO "audio support initialized\n");
 		platform_set_drvdata(pdev, card);
 		return 0;
 	}
-	
+
 nodev3:
-	kfree(alsa_codec);	
-nodev2:	
+	kfree(alsa_codec);
+nodev2:
 	snd_card_free(card);
 nodev1:
 	omap_mcbsp_stop(AUDIO_MCBSP);
@@ -567,14 +576,14 @@ int snd_omap_alsa_remove(struct platform_device *pdev)
 {
 	struct snd_card *card = platform_get_drvdata(pdev);
 	struct snd_card_omap_codec *chip = card->private_data;
-	
+
 	snd_card_free(card);
 
 	alsa_codec = NULL;
 	card->private_data = NULL;
 	kfree(chip);
-	
+
 	platform_set_drvdata(pdev, NULL);
-	
+
 	return 0;
 }
-- 
1.5.5-rc3.GIT

Re: [PATCH 00/19] Update and clean up on sound/arm/omap/omap-alsa*[c,h] (take #2)

[Tony Lindgren]

* Eduardo Valentin <edubezval@gmail.com> [080418 01:03]:
> From: Eduardo Valentin <eduardo.valentin@indt.org.br>
> 
> Hi all,
> 
> This patch series cleans up most of the errors and warnings reported
> by checkpatch.pl on sound/arm/omap/omap-alsa*[c,h]. Some checks reported
> by checkpatch.pl were ignored, such as removing #if 0 codes. Not all were
> eligible for remotion.
> 
> There is also one patch that updates the alsa driver for H2.
> Since tsc2101 i2c driver was updated, this alsa driver stop
> working.

Pushing this today, you might want to talk with Jarkko Nikula on
converting things to use the asoc driver :)

Tony


> 
> Cheers,
> 
> ---
> Eduardo Valentin
> 
> Eduardo Valentin (19):
>   Update audio driver for H2 board
>   Code clean-up for include/asm-arm/arch-omap/omap-alsa.h
>   Code clean-up for sound/arm/omap/omap-alsa-aic23.c
>   Code clean-up for sound/arm/omap/omap-alsa-aic23.h
>   Code clean-up for sound/arm/omap/omap-alsa-aic23-mixer.c
>   Code clean-up for sound/arm/omap/omap-alsa-dma.c
>   Code clean-up for sound/arm/omap/omap-alsa-dma.h
>   Code clean-up for sound/arm/omap/omap-alsa-sx1-mixer.c
>   Code clean-up for sound/arm/omap/omap-alsa-sx1-mixer.h
>   Code clean-up for sound/arm/omap/omap-alsa-sx1.c
>   Code clean-up for sound/arm/omap/omap-alsa-sx1.h
>   Code clean-up for sound/arm/omap/omap-alsa-tsc2101-mixer.c
>   Code clean-up for sound/arm/omap/omap-alsa-tsc2101-mixer.h
>   Code clean-up for sound/arm/omap/omap-alsa-tsc2101.c
>   Code clean-up for sound/arm/omap/omap-alsa-tsc2101.h
>   Code clean-up for sound/arm/omap/omap-alsa-tsc2102-mixer.c
>   Code clean-up for sound/arm/omap/omap-alsa-tsc2102.c
>   Code clean-up for sound/arm/omap/omap-alsa-tsc2102.h
>   Code clean-up for sound/arm/omap/omap-alsa.c
> 
>  arch/arm/mach-omap1/board-h2.c           |   71 ++--
>  include/asm-arm/arch-omap/omap-alsa.h    |   29 +-
>  sound/arm/omap/omap-alsa-aic23-mixer.c   |  144 ++++---
>  sound/arm/omap/omap-alsa-aic23.c         |   59 ++--
>  sound/arm/omap/omap-alsa-aic23.h         |   12 +-
>  sound/arm/omap/omap-alsa-dma.c           |  171 ++++-----
>  sound/arm/omap/omap-alsa-dma.h           |   22 +-
>  sound/arm/omap/omap-alsa-sx1-mixer.c     |   22 +-
>  sound/arm/omap/omap-alsa-sx1-mixer.h     |    6 +-
>  sound/arm/omap/omap-alsa-sx1.c           |   72 +++-
>  sound/arm/omap/omap-alsa-sx1.h           |    3 +-
>  sound/arm/omap/omap-alsa-tsc2101-mixer.c |  631 ++++++++++++++++--------------
>  sound/arm/omap/omap-alsa-tsc2101-mixer.h |   31 +-
>  sound/arm/omap/omap-alsa-tsc2101.c       |  320 ++++++++++-----
>  sound/arm/omap/omap-alsa-tsc2101.h       |   11 +-
>  sound/arm/omap/omap-alsa-tsc2102-mixer.c |   14 +-
>  sound/arm/omap/omap-alsa-tsc2102.c       |   70 ++--
>  sound/arm/omap/omap-alsa-tsc2102.h       |    2 +-
>  sound/arm/omap/omap-alsa.c               |  163 ++++----
>  19 files changed, 1056 insertions(+), 797 deletions(-)
> 
> --
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