Merge source.mvista.com:linux-omap

arch/arm/mach-omap1/board-h2.c

@@ -40,8 +40,9 @@
 #include <asm/arch/irda.h>
 #include <asm/arch/usb.h>
 #include <asm/arch/keypad.h>
-#include <asm/arch/dma.h>
 #include <asm/arch/common.h>
+#include <asm/arch/mcbsp.h>
+#include <asm/arch/omap-alsa.h>
 
 extern int omap_gpio_init(void);
 
@@ -285,6 +286,41 @@ static struct platform_device h2_lcd_device = {
 	.id		= -1,
 };
 
+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,
@@ -292,6 +328,7 @@ static struct platform_device *h2_devices[] __initdata = {
 	&h2_irda_device,
 	&h2_kp_device,
 	&h2_lcd_device,
+	&h2_mcbsp1_device,
 };
 
 static void __init h2_init_smc91x(void)

arch/arm/mach-omap1/board-osk.c

@@ -47,6 +47,8 @@
 #include <asm/arch/tc.h>
 #include <asm/arch/keypad.h>
 #include <asm/arch/common.h>
+#include <asm/arch/mcbsp.h>
+#include <asm/arch/omap-alsa.h>
 
 static int osk_keymap[] = {
 	KEY(0, 0, KEY_F1),
@@ -149,9 +151,40 @@ static struct platform_device osk5912_cf_device = {
 	.resource	= osk5912_cf_resources,
 };
 
+#define DEFAULT_BITPERSAMPLE 16
+
+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(0),
+	.rcr1 = RFRLEN1(OMAP_MCBSP_WORD_8) | RWDLEN1(OMAP_MCBSP_WORD_16),
+	.xcr2 = XPHASE | XFRLEN2(OMAP_MCBSP_WORD_8) |
+	    XWDLEN2(OMAP_MCBSP_WORD_16) | XDATDLY(0) | XFIG,
+	.xcr1 = XFRLEN1(OMAP_MCBSP_WORD_8) | XWDLEN1(OMAP_MCBSP_WORD_16),
+	.srgr1 = FWID(DEFAULT_BITPERSAMPLE - 1),
+	.srgr2 = GSYNC | CLKSP | FSGM | FPER(DEFAULT_BITPERSAMPLE * 2 - 1),
+	/*.pcr0 = FSXM | FSRM | CLKXM | CLKRM | CLKXP | CLKRP,*/ /* mcbsp: master */
+	.pcr0 = CLKXP | CLKRP,  /* mcbsp: slave */
+};
+
+static struct omap_alsa_codec_config alsa_config = {
+	.name			= "OSK AIC23",
+	.mcbsp_regs_alsa	= &mcbsp_regs,
+	.codec_configure_dev	= NULL, // aic23_configure,
+	.codec_set_samplerate	= NULL, // aic23_set_samplerate,
+	.codec_clock_setup	= NULL, // aic23_clock_setup,
+	.codec_clock_on		= NULL, // aic23_clock_on,
+	.codec_clock_off	= NULL, // aic23_clock_off,
+	.get_default_samplerate	= NULL, // aic23_get_default_samplerate,	
+};
+
 static struct platform_device osk5912_mcbsp1_device = {
-	.name		= "omap_mcbsp",
-	.id		= 1,
+	.name	= "omap_alsa_mcbsp",
+ 	.id	= 1,
+	.dev = {
+		.platform_data	= &alsa_config,
+	},
 };
 
 static struct resource osk5912_kp_resources[] = {

drivers/cbus/tahvo-usb.c

@@ -193,8 +193,9 @@ static int omap_otg_probe(struct device *dev)
 
 static int omap_otg_remove(struct device *dev)
 {
-	tahvo_otg_dev = NULL;
 	free_irq(tahvo_otg_dev->resource[1].start, &tahvo_usb_device);
+	tahvo_otg_dev = NULL;
+
 	return 0;
 }
 

sound/arm/omap-aic23.h → include/asm-arm/arch-omap/omap-alsa.h

 /*
- * sound/arm/omap-aic23.h
+ * linux/include/asm-arm/arch-omap/omap-alsa.h
  * 
- * Alsa Driver for AIC23 codec on OSK5912 platform board
+ * Alsa Driver for AIC23 and TSC2101 codecs on OMAP platform boards.
+ *
+ * Copyright (C) 2006 Mika Laitio <lamikr@cc.jyu.fi>
  *
  * Copyright (C) 2005 Instituto Nokia de Tecnologia - INdT - Manaus Brazil
  * Written by Daniel Petrini, David Cohen, Anderson Briglia
@@ -33,39 +35,38 @@
  *  2005/07/25 INdT-10LE Kernel Team - 	Alsa driver for omap osk,
  *  					original version based in sa1100 driver
  *  					and omap oss driver.
- *
- *  2005-12-18   Dirk Behme      - Added L/R Channel Interchange fix as proposed by Ajaya Babu
  */
 
-#ifndef __OMAP_AIC23_H
-#define __OMAP_AIC23_H
+#ifndef __OMAP_ALSA_H
+#define __OMAP_ALSA_H
 
 #include <sound/driver.h>
 #include <asm/arch/dma.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
+#include <asm/arch/mcbsp.h>
+#include <linux/platform_device.h>
+/*
+ * Debug functions
+ */
+#undef DEBUG
+//#define DEBUG
+
+#define ERR(ARGS...) printk(KERN_ERR "{%s}-ERROR: ", __FUNCTION__);printk(ARGS);
+
+#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)
+#else
+#define DPRINTK(ARGS...)	/* nop */
+#define ADEBUG()		/* nop */
+#define FN_IN			/* nop */
+#define FN_OUT(n)		/* nop */
+#endif
 
-#define DEFAULT_OUTPUT_VOLUME         0x60
-#define DEFAULT_INPUT_VOLUME          0x00	/* 0 ==> mute line in */
-
-#define OUTPUT_VOLUME_MIN             LHV_MIN
-#define OUTPUT_VOLUME_MAX             LHV_MAX
-#define OUTPUT_VOLUME_RANGE           (OUTPUT_VOLUME_MAX - OUTPUT_VOLUME_MIN)
-#define OUTPUT_VOLUME_MASK            OUTPUT_VOLUME_MAX
-
-#define INPUT_VOLUME_MIN 	      LIV_MIN
-#define INPUT_VOLUME_MAX 	      LIV_MAX
-#define INPUT_VOLUME_RANGE 	      (INPUT_VOLUME_MAX - INPUT_VOLUME_MIN)
-#define INPUT_VOLUME_MASK 	      INPUT_VOLUME_MAX
-
-#define SIDETONE_MASK                 0x1c0
-#define SIDETONE_0                    0x100
-#define SIDETONE_6                    0x000
-#define SIDETONE_9                    0x040
-#define SIDETONE_12                   0x080
-#define SIDETONE_18                   0x0c0
-
-#define DEFAULT_ANALOG_AUDIO_CONTROL  DAC_SELECTED | STE_ENABLED | BYPASS_ON | INSEL_MIC | MICB_20DB
+#define DMA_BUF_SIZE	(1024 * 8)
 
 /*
  * Buffer management for alsa and dma
@@ -93,38 +94,51 @@ struct audio_stream {
 /*
  * Alsa card structure for aic23
  */
-struct snd_card_omap_aic23 {
+struct snd_card_omap_codec {
 	snd_card_t *card;
 	snd_pcm_t *pcm;
 	long samplerate;
 	struct audio_stream s[2];	/* playback & capture */
 };
 
-/*********** Function Prototypes *************************/
+/* Codec specific information and function pointers. 
+ * Codec (omap-alsa-aic23.c and omap-alsa-tsc2101.c)
+ * are responsible for defining the function pointers.
+ */
+struct omap_alsa_codec_config {
+	char 	*name;
+	struct	omap_mcbsp_reg_cfg *mcbsp_regs_alsa;
+	snd_pcm_hw_constraint_list_t *hw_constraints_rates;
+	snd_pcm_hardware_t *snd_omap_alsa_playback;
+	snd_pcm_hardware_t *snd_omap_alsa_capture;
+	void	(*codec_configure_dev)(void);
+	void	(*codec_set_samplerate)(long);
+	void	(*codec_clock_setup)(void);
+	int	(*codec_clock_on)(void);
+	int 	(*codec_clock_off)(void);
+	int	(*get_default_samplerate)(void);
+};
 
-void audio_dma_callback(void *);
-int snd_omap_mixer(struct snd_card_omap_aic23 *);
+/*********** Mixer function prototypes *************************/
+int snd_omap_mixer(struct snd_card_omap_codec *);
 void snd_omap_init_mixer(void);
-/* Clock functions */
-int omap_aic23_clock_on(void);
-int omap_aic23_clock_off(void);
 
 #ifdef CONFIG_PM
 void snd_omap_suspend_mixer(void);
 void snd_omap_resume_mixer(void);
 #endif
 
-/* Codec AIC23 */
-#if defined(CONFIG_SENSORS_TLV320AIC23) || defined (CONFIG_SENSORS_TLV320AIC23_MODULE)
-
-extern int tlv320aic23_write_value(u8 reg, u16 value);
-
-/* TLV320AIC23 is a write only device */
-static __inline__ void audio_aic23_write(u8 address, u16 data)
-{
-	tlv320aic23_write_value(address, data);
-}
+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);
+int snd_omap_alsa_resume(struct platform_device *pdev);
+#else
+#define snd_omap_alsa_suspend	NULL
+#define snd_omap_alsa_resume	NULL
+#endif
 
-#endif /* CONFIG_SENSORS_TLV320AIC23 */
+/*********** function prototype to function called from the dma interrupt handler ******/
+void callback_omap_alsa_sound_dma(void *);
 
 #endif

sound/arm/Kconfig

@@ -44,5 +44,19 @@ config SND_OMAP_AIC23
 
 	  To compile this driver as a module, choose M here: the module
 	  will be called snd-omap-aic23.
+	  
+config SND_OMAP_TSC2101
+ 	tristate "OMAP TSC2101 alsa driver"
+ 	depends on ARCH_OMAP && SND
+ 	select SND_PCM
+         select OMAP_TSC2101
+         select OMAP_UWIRE if ARCH_OMAP
+ 	help
+ 	  Say Y here if you have a OMAP platform board
+ 	  and want to use its TSC2101 audio chip. Driver has
+ 	  been tested with H2 and iPAQ h6300.
+ 
+ 	  To compile this driver as a module, choose M here: the module
+ 	  will be called snd-omap-tsc2101.
 
 endmenu

sound/arm/Makefile

@@ -14,5 +14,4 @@ snd-pxa2xx-pcm-objs		:= pxa2xx-pcm.o
 obj-$(CONFIG_SND_PXA2XX_AC97)	+= snd-pxa2xx-ac97.o
 snd-pxa2xx-ac97-objs		:= pxa2xx-ac97.o
 
-obj-$(CONFIG_SND_OMAP_AIC23) += snd-omap-aic23.o
-snd-omap-aic23-objs := omap-aic23.o omap-alsa-dma.o omap-alsa-mixer.o
+obj-$(CONFIG_SND) += omap/

sound/arm/omap/Makefile

+#
+## Makefile for ALSA OMAP
+#
+#
+obj-$(CONFIG_SND_OMAP_AIC23) += snd-omap-alsa-aic23.o
+snd-omap-alsa-aic23-objs := omap-alsa.o omap-alsa-dma.o omap-alsa-aic23.o omap-alsa-aic23-mixer.o
+
+obj-$(CONFIG_SND_OMAP_TSC2101) += snd-omap-alsa-tsc2101.o
+snd-omap-alsa-tsc2101-objs := omap-alsa.o omap-alsa-dma.o omap-alsa-tsc2101.o omap-alsa-tsc2101-mixer.o

sound/arm/omap-alsa-mixer.c → sound/arm/omap/omap-alsa-aic23-mixer.c

 /*
- * sound/arm/omap-alsa-mixer.c
+ * sound/arm/omap/omap-alsa-aic23-mixer.c
  * 
  * Alsa Driver Mixer for generic codecs for omap boards
  *
@@ -39,20 +39,10 @@
 
 #include <linux/config.h>
 #include <sound/driver.h>
-#include <linux/module.h>
-#include <linux/device.h>
-#include <linux/init.h>
-#include <linux/errno.h>
-#include <linux/ioctl.h>
-#include <linux/delay.h>
-#include <linux/slab.h>
-
-#include <asm/hardware.h>
-#include <asm/mach-types.h>
-#include <asm/arch/dma.h>
 #include <asm/arch/aic23.h>
 
-#include "omap-aic23.h"
+#include <asm/arch/omap-alsa.h>
+#include "omap-alsa-aic23.h"
 #include <sound/initval.h>
 #include <sound/control.h>
 
@@ -67,7 +57,7 @@ MODULE_DESCRIPTION("OMAP Alsa mixer driver for ALSA");
 /* Codec AIC23 */
 #if defined(CONFIG_SENSORS_TLV320AIC23) || defined (CONFIG_SENSORS_TLV320AIC23_MODULE)
 
-extern __inline__ void audio_aic23_write(u8, u16);
+extern void audio_aic23_write(u8, u16);
 
 #define MIXER_NAME		     "Mixer AIC23"
 #define SND_OMAP_WRITE(reg, val)     audio_aic23_write(reg, val)
@@ -411,31 +401,6 @@ static snd_kcontrol_new_t snd_omap_controls[] = {
 	OMAP_MUX("Capture Source", ANALOG_AUDIO_CONTROL_ADDR, AAC_INDEX, INSEL_MIC),
 };
 
-void snd_omap_init_mixer(void)
-{
-	u16 vol_reg;
-
-	/* Line's default values */
-	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);
-	
-	/* Analog Audio Control's default values */
-	omap_regs[AAC_INDEX].l_reg = DEFAULT_ANALOG_AUDIO_CONTROL;
-	
-	/* Headphone's default values */
-	vol_reg = LZC_ON;
-	vol_reg &= ~OUTPUT_VOLUME_MASK;
-	vol_reg |= DEFAULT_OUTPUT_VOLUME;
-	omap_regs[PCM_INDEX].l_reg = DEFAULT_OUTPUT_VOLUME;
-	omap_regs[PCM_INDEX].r_reg = DEFAULT_OUTPUT_VOLUME;
-	omap_regs[PCM_INDEX].sw = 1;
-	SND_OMAP_WRITE(LEFT_CHANNEL_VOLUME_ADDR, vol_reg);
-	SND_OMAP_WRITE(RIGHT_CHANNEL_VOLUME_ADDR, vol_reg);
-}
-
 #ifdef CONFIG_PM
 
 void snd_omap_suspend_mixer(void)
@@ -474,7 +439,32 @@ void snd_omap_resume_mixer(void)
 }
 #endif
 
-int snd_omap_mixer(struct snd_card_omap_aic23 *chip)
+void snd_omap_init_mixer(void)
+{
+	u16 vol_reg;
+
+	/* Line's default values */
+	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);
+	
+	/* Analog Audio Control's default values */
+	omap_regs[AAC_INDEX].l_reg = DEFAULT_ANALOG_AUDIO_CONTROL;
+	
+	/* Headphone's default values */
+	vol_reg = LZC_ON;
+	vol_reg &= ~OUTPUT_VOLUME_MASK;
+	vol_reg |= DEFAULT_OUTPUT_VOLUME;
+	omap_regs[PCM_INDEX].l_reg = DEFAULT_OUTPUT_VOLUME;
+	omap_regs[PCM_INDEX].r_reg = DEFAULT_OUTPUT_VOLUME;
+	omap_regs[PCM_INDEX].sw = 1;
+	SND_OMAP_WRITE(LEFT_CHANNEL_VOLUME_ADDR, vol_reg);
+	SND_OMAP_WRITE(RIGHT_CHANNEL_VOLUME_ADDR, vol_reg);
+}
+
+int snd_omap_mixer(struct snd_card_omap_codec *chip)
 {
 	snd_card_t *card;
 	unsigned int idx;
@@ -493,4 +483,3 @@ int snd_omap_mixer(struct snd_card_omap_aic23 *chip)
 
 	return 0;
 }
-

sound/arm/omap/omap-alsa-aic23.c

+/*
+ * 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
+ * Written by Daniel Petrini, David Cohen, Anderson Briglia
+ *            {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
+ * 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
+ * option) any later version.
+ */
+
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/control.h>
+#include <linux/clk.h>
+#include <asm/arch/clock.h>
+#include <asm/arch/aic23.h>
+
+#include <asm/arch/omap-alsa.h>
+#include "omap-alsa-aic23.h"
+
+static struct clk *aic23_mclk = 0;
+
+/* aic23 related */
+static const struct aic23_samplerate_reg_info
+ rate_reg_info[NUMBER_SAMPLE_RATES_SUPPORTED] = {
+	{4000, 0x06, 1},		/*  4000 */
+	{8000, 0x06, 0},		/*  8000 */
+	{16000, 0x0C, 1},		/* 16000 */
+	{22050, 0x11, 1},               /* 22050 */
+	{24000, 0x00, 1},		/* 24000 */
+	{32000, 0x0C, 0},		/* 32000 */
+	{44100, 0x11, 0},		/* 44100 */
+	{48000, 0x00, 0},		/* 48000 */
+	{88200, 0x1F, 0},		/* 88200 */
+	{96000, 0x0E, 0},		/* 96000 */
+};
+
+/*
+ * Hardware capabilities
+ */
+ 
+ /*
+ * DAC USB-mode sampling rates (MCLK = 12 MHz)
+ * The rates and rate_reg_into MUST be in the same order
+ */
+static unsigned int rates[] = {
+	4000, 8000, 16000, 22050,
+	24000, 32000, 44100,
+	48000, 88200, 96000,
+};
+
+static snd_pcm_hw_constraint_list_t aic23_hw_constraints_rates = {
+	.count = ARRAY_SIZE(rates),
+	.list = rates,
+	.mask = 0,
+};
+
+static snd_pcm_hardware_t aic23_snd_omap_alsa_playback = {
+	.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_16000 |
+		  SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |
+		  SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
+		  SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
+		  SNDRV_PCM_RATE_KNOT),
+	.rate_min = 8000,
+	.rate_max = 96000,
+	.channels_min = 2,
+	.channels_max = 2,
+	.buffer_bytes_max = 128 * 1024,
+	.period_bytes_min = 32,
+	.period_bytes_max = 8 * 1024,
+	.periods_min = 16,
+	.periods_max = 255,
+	.fifo_size = 0,
+};
+
+static snd_pcm_hardware_t aic23_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_16000 |
+		  SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |
+		  SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
+		  SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
+		  SNDRV_PCM_RATE_KNOT),
+	.rate_min = 8000,
+	.rate_max = 96000,
+	.channels_min = 2,
+	.channels_max = 2,
+	.buffer_bytes_max = 128 * 1024,
+	.period_bytes_min = 32,
+	.period_bytes_max = 8 * 1024,
+	.periods_min = 16,
+	.periods_max = 255,
+	.fifo_size = 0,
+};
+
+/*
+ * Codec/mcbsp init and configuration section
+ * codec dependent code.
+ */
+
+extern int tlv320aic23_write_value(u8 reg, u16 value);
+
+/* TLV320AIC23 is a write only device */
+void audio_aic23_write(u8 address, u16 data)
+{
+	tlv320aic23_write_value(address, data);
+}
+EXPORT_SYMBOL_GPL(audio_aic23_write);
+
+/*
+ * Sample rate changing
+ */
+void aic23_set_samplerate(long rate)
+{
+	u8 count = 0;
+	u16 data = 0;
+
+	/* Fix the rate if it has a wrong value */
+	if (rate >= 96000)
+		rate = 96000;
+	else if (rate >= 88200)
+		rate = 88200;
+	else if (rate >= 48000)
+		rate = 48000;
+	else if (rate >= 44100)
+		rate = 44100;
+	else if (rate >= 32000)
+		rate = 32000;
+	else if (rate >= 24000)
+		rate = 24000;
+	else if (rate >= 22050)
+		rate = 22050;
+	else if (rate >= 16000)
+		rate = 16000;
+	else if (rate >= 8000)
+		rate = 8000;
+	else
+		rate = 4000;
+
+	/* Search for the right sample rate */
+	/* Verify what happens if the rate is not supported
+	 * now it goes to 96Khz */
+	while ((rate_reg_info[count].sample_rate != rate) &&
+	       (count < (NUMBER_SAMPLE_RATES_SUPPORTED - 1))) {
+		count++;
+	}
+
+	data = (rate_reg_info[count].divider << CLKIN_SHIFT) |
+	    (rate_reg_info[count].control << BOSR_SHIFT) | USB_CLK_ON;
+
+	audio_aic23_write(SAMPLE_RATE_CONTROL_ADDR, data);
+}
+
+inline void aic23_configure(void)
+{
+	/* Reset codec */
+	audio_aic23_write(RESET_CONTROL_ADDR, 0);
+
+	/* 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);
+
+	/* Digital audio path control, de-emphasis control 44.1kHz */
+	audio_aic23_write(DIGITAL_AUDIO_CONTROL_ADDR, DEEMP_44K);
+
+	/* Digital audio interface, master/slave mode, I2S, 16 bit */
+#ifdef AIC23_MASTER
+	audio_aic23_write(DIGITAL_AUDIO_FORMAT_ADDR,
+			  MS_MASTER | IWL_16 | FOR_DSP);
+#else
+	audio_aic23_write(DIGITAL_AUDIO_FORMAT_ADDR, IWL_16 | FOR_DSP);
+#endif
+
+	/* Enable digital interface */
+	audio_aic23_write(DIGITAL_INTERFACE_ACT_ADDR, ACT_ON);
+}
+
+/*
+ *  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. 
+ */
+/*
+ * Do clock framework mclk search
+ */
+void aic23_clock_setup(void)
+{
+	aic23_mclk = clk_get(0, "mclk");
+}
+
+/*
+ * Do some sanity check, set clock rate, starts it and
+ *  turn codec audio on 
+ */
+int aic23_clock_on(void)
+{
+	if (clk_get_usecount(aic23_mclk) > 0) {
+		/* MCLK is already in use */
+		printk(KERN_WARNING
+		       "MCLK in use at %d Hz. We change it to %d Hz\n",
+		       (uint) clk_get_rate(aic23_mclk),
+		       CODEC_CLOCK);
+	}
+	
+	if (clk_set_rate(aic23_mclk, CODEC_CLOCK)) {
+		printk(KERN_ERR
+		       "Cannot set MCLK for AIC23 CODEC\n");
+		return -ECANCELED;
+	}
+
+	clk_enable(aic23_mclk);
+
+	printk(KERN_DEBUG
+		"MCLK = %d [%d], usecount = %d\n",
+	       (uint) clk_get_rate(aic23_mclk), CODEC_CLOCK,
+	       clk_get_usecount(aic23_mclk));
+
+	/* Now turn the audio on */
+	audio_aic23_write(POWER_DOWN_CONTROL_ADDR, 
+			  ~DEVICE_POWER_OFF & ~OUT_OFF & ~DAC_OFF &
+			  ~ADC_OFF & ~MIC_OFF & ~LINE_OFF);	
+	return 0;
+}
+/*
+ * Do some sanity check, turn clock off and then turn
+ *  codec audio off
+ */
+int aic23_clock_off(void)
+{
+	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",
+			       (uint) clk_get_rate(aic23_mclk),
+			       CODEC_CLOCK);
+		}
+
+		clk_disable(aic23_mclk);
+	}
+	
+	audio_aic23_write(POWER_DOWN_CONTROL_ADDR,
+			  DEVICE_POWER_OFF | OUT_OFF | DAC_OFF |
+			  ADC_OFF | MIC_OFF | LINE_OFF);	
+	return 0;
+}
+
+int aic23_get_default_samplerate(void)
+{
+	return DEFAULT_SAMPLE_RATE;
+}
+
+static int __init 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->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;
+		ret	= snd_omap_alsa_post_probe(pdev, codec_cfg);
+	}
+	else
+		ret = -ENODEV;
+	return ret;
+}
+
+static struct platform_driver omap_alsa_driver = {
+	.probe		= snd_omap_alsa_aic23_probe,
+	.remove 	= snd_omap_alsa_remove,
+	.suspend	= snd_omap_alsa_suspend,
+	.resume		= snd_omap_alsa_resume,
+	.driver	= {
+		.name =	"omap_alsa_mcbsp",
+	},
+};
+
+static int __init omap_alsa_aic23_init(void)
+{
+	int err;
+	
+	ADEBUG();
+	err = platform_driver_register(&omap_alsa_driver);
+
+	return err;
+}
+
+static void __exit omap_alsa_aic23_exit(void)
+{
+	ADEBUG();
+	
+	platform_driver_unregister(&omap_alsa_driver);
+}
+
+module_init(omap_alsa_aic23_init);
+module_exit(omap_alsa_aic23_exit);

sound/arm/omap/omap-alsa-aic23.h

+/*
+ * 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
+ * Written by Daniel Petrini, David Cohen, Anderson Briglia
+ *            {daniel.petrini, david.cohen, anderson.briglia}@indt.org.br
+ *
+ * Copyright (C) 2006 Mika Laitio <lamikr@cc.jyu.fi>
+ *
+ * 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
+ * option) any later version.
+ */
+
+#ifndef __OMAP_ALSA_AIC23_H
+#define __OMAP_ALSA_AIC23_H
+
+#include <sound/driver.h>
+#include <asm/arch/dma.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <asm/arch/mcbsp.h>
+
+/* Define to set the AIC23 as the master w.r.t McBSP */
+#define AIC23_MASTER
+
+#define NUMBER_SAMPLE_RATES_SUPPORTED	10
+
+/*
+ * AUDIO related MACROS
+ */
+#ifndef DEFAULT_BITPERSAMPLE
+#define DEFAULT_BITPERSAMPLE		16
+#endif
+
+#define DEFAULT_SAMPLE_RATE		44100
+#define CODEC_CLOCK			12000000
+#define AUDIO_MCBSP			OMAP_MCBSP1
+
+#define DEFAULT_OUTPUT_VOLUME		0x60
+#define DEFAULT_INPUT_VOLUME		0x00	/* 0 ==> mute line in */
+
+#define OUTPUT_VOLUME_MIN		LHV_MIN
+#define OUTPUT_VOLUME_MAX		LHV_MAX
+#define OUTPUT_VOLUME_RANGE		(OUTPUT_VOLUME_MAX - OUTPUT_VOLUME_MIN)
+#define OUTPUT_VOLUME_MASK		OUTPUT_VOLUME_MAX
+
+#define INPUT_VOLUME_MIN		LIV_MIN
+#define INPUT_VOLUME_MAX		LIV_MAX
+#define INPUT_VOLUME_RANGE		(INPUT_VOLUME_MAX - INPUT_VOLUME_MIN)
+#define INPUT_VOLUME_MASK		INPUT_VOLUME_MAX
+
+#define SIDETONE_MASK			0x1c0
+#define SIDETONE_0			0x100
+#define SIDETONE_6			0x000
+#define SIDETONE_9			0x040
+#define SIDETONE_12			0x080
+#define SIDETONE_18			0x0c0
+
+#define DEFAULT_ANALOG_AUDIO_CONTROL  DAC_SELECTED | STE_ENABLED | BYPASS_ON | INSEL_MIC | MICB_20DB
+
+struct aic23_samplerate_reg_info {
+	u32 sample_rate;
+	u8 control;		/* SR3, SR2, SR1, SR0 and BOSR */
+	u8 divider;		/* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */
+};
+
+/*
+ * Defines codec specific functions pointers that can be used from the 
+ * common omap-alse 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);
+void aic23_set_samplerate(long rate);
+void aic23_clock_setup(void);
+int aic23_clock_on(void);
+int aic23_clock_off(void);
+int aic23_get_default_samplerate(void);
+
+#endif

sound/arm/omap-alsa-dma.c → sound/arm/omap/omap-alsa-dma.c

 /*
- * sound/arm/omap-alsa-dma.c
+ * sound/arm/omap/omap-alsa-dma.c
  *
  * Common audio DMA handling for the OMAP processors
  *
+ * 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.
@@ -66,20 +68,9 @@
 
 #include <asm/arch/mcbsp.h>
 
-#include "omap-aic23.h"
+#include <asm/arch/omap-alsa.h>
 
 #undef DEBUG
-//#define DEBUG
-#ifdef DEBUG
-#define DPRINTK(ARGS...)  printk(KERN_INFO "<%s>: ",__FUNCTION__);printk(ARGS)
-#define FN_IN printk(KERN_INFO "[%s]: start\n", __FUNCTION__)
-#define FN_OUT(n) printk(KERN_INFO "[%s]: end(%u)\n",__FUNCTION__, n)
-#else
-
-#define DPRINTK( x... )
-#define FN_IN
-#define FN_OUT(x)
-#endif
 
 #define ERR(ARGS...) printk(KERN_ERR "{%s}-ERROR: ", __FUNCTION__);printk(ARGS);
 
@@ -165,7 +156,7 @@ static void omap_sound_dma_link_lch(void *data)
 	FN_OUT(0);
 }
 
-int omap_request_sound_dma(int device_id, const char *device_name,
+int omap_request_alsa_sound_dma(int device_id, const char *device_name,
 			   void *data, int **channels)
 {
 	int i, err = 0;
@@ -186,10 +177,11 @@ int omap_request_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, device_name,
-				     sound_dma_irq_handler, data,
-				     &chan[i]);
+		err = omap_request_dma(device_id, 
+				device_name,
+				sound_dma_irq_handler, 
+				data,
+				&chan[i]);
 
 		/* Handle Failure condition here */
 		if (err < 0) {
@@ -223,7 +215,7 @@ int omap_request_sound_dma(int device_id, const char *device_name,
  **************************************************************************************/
 static void omap_sound_dma_unlink_lch(void *data)
 {
-	struct audio_stream *s = (struct audio_stream *) data;
+	struct audio_stream *s = (struct audio_stream *)data;
 	int *chan = s->lch;
 	int i;
 
@@ -243,11 +235,11 @@ static void omap_sound_dma_unlink_lch(void *data)
 	FN_OUT(0);
 }
 
-int omap_free_sound_dma(void *data, int **channels)
+int omap_free_alsa_sound_dma(void *data, int **channels)
 {
-
 	int i;
 	int *chan = NULL;
+	
 	FN_IN;
 	if (unlikely(NULL == channels)) {
 		BUG();
@@ -277,10 +269,11 @@ int omap_free_sound_dma(void *data, int **channels)
  * Stop all the DMA channels of the stream
  *
  **************************************************************************************/
-void omap_audio_stop_dma(struct audio_stream *s)
+void omap_stop_alsa_sound_dma(struct audio_stream *s)
 {
 	int *chan = s->lch;
 	int i;
+	
 	FN_IN;
 	if (unlikely(NULL == chan)) {
 		BUG();
@@ -299,7 +292,7 @@ void omap_audio_stop_dma(struct audio_stream *s)
  * Clear any pending transfers
  *
  **************************************************************************************/
-void omap_clear_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]);
@@ -307,13 +300,6 @@ void omap_clear_sound_dma(struct audio_stream * s)
 	return;
 }
 
-/*********************************** MODULE FUNCTIONS DEFINTIONS ***********************/
-
-#ifdef OMAP1610_MCBSP1_BASE
-#undef OMAP1610_MCBSP1_BASE
-#endif
-#define OMAP1610_MCBSP1_BASE    0xE1011000
-
 /***************************************************************************************
  *
  * DMA related functions
@@ -325,9 +311,10 @@ static int audio_set_dma_params_play(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_dest_params(channel, 0x05, 0x00,
-				 (OMAP1610_MCBSP1_BASE + 0x806),
+				 (OMAP1510_MCBSP1_BASE + 0x06),
 				 0, 0);
 	omap_set_dma_src_params(channel, 0x00, 0x01, dma_ptr,
 				0, 0);
@@ -341,11 +328,11 @@ 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,
-				(OMAP1610_MCBSP1_BASE + 0x802),
+				(OMAP1510_MCBSP1_BASE + 0x02),
 				0, 0);
 	omap_set_dma_dest_params(channel, 0x00, 0x01, dma_ptr, 0, 0);
 	omap_set_dma_transfer_params(channel, dt, cen, cfn, 0x00, 0, 0);
@@ -358,7 +345,7 @@ 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 */
@@ -372,8 +359,9 @@ static int audio_start_dma_chain(struct audio_stream *s)
  * Do the initial set of work to initialize all the channels as required.
  * We shall then initate a transfer
  */
-int omap_start_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)
 {
 	int ret = -EPERM;
 
@@ -439,18 +427,17 @@ static void sound_dma_irq_handler(int sound_curr_lch, u16 ch_status,
 	}
 
 	if (ch_status & DCSR_END_BLOCK) 
-		audio_dma_callback(s);
+		callback_omap_alsa_sound_dma(s);
 	FN_OUT(0);
 	return;
 }
 
 MODULE_AUTHOR("Texas Instruments");
-MODULE_DESCRIPTION
-    ("Common DMA handling for Audio driver on OMAP processors");
+MODULE_DESCRIPTION("Common DMA handling for Audio driver on OMAP processors");
 MODULE_LICENSE("GPL");
 
-EXPORT_SYMBOL(omap_start_sound_dma);
-EXPORT_SYMBOL(omap_clear_sound_dma);
-EXPORT_SYMBOL(omap_request_sound_dma);
-EXPORT_SYMBOL(omap_free_sound_dma);
-EXPORT_SYMBOL(omap_audio_stop_dma);
+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);

sound/arm/omap-alsa-dma.h → sound/arm/omap/omap-alsa-dma.h

 /*  
- * linux/sound/arm/omap-alsa-dma.h
+ * linux/sound/arm/omap/omap-alsa-dma.h
  *
  * Common audio DMA handling for the OMAP processors
  *
+ * 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.
@@ -30,14 +32,7 @@
 
 /************************** INCLUDES *************************************/
 
-#include "omap-aic23.h"
-
-/************************** GLOBAL MACROS *************************************/
-
-/* Provide the Macro interfaces common across platforms */
-#define DMA_REQUEST(e,s, cb)	{e=omap_request_sound_dma(s->dma_dev, s->id, s, &s->lch);}
-#define DMA_FREE(s)		omap_free_sound_dma(s, &s->lch)
-#define DMA_CLEAR(s)		omap_clear_sound_dma(s)
+#include <asm/arch/omap-alsa.h>
 
 /************************** GLOBAL DATA STRUCTURES *********************************/
 
@@ -45,15 +40,14 @@ typedef void (*dma_callback_t) (int lch, u16 ch_status, void *data);
 
 /**************** ARCH SPECIFIC FUNCIONS *******************************************/
 
-void omap_clear_sound_dma(struct audio_stream * s);
+void omap_clear_alsa_sound_dma(struct audio_stream * s);
 
-int omap_request_sound_dma(int device_id, const char *device_name,
+int omap_request_alsa_sound_dma(int device_id, const char *device_name,
 			   void *data, int **channels);
-int omap_free_sound_dma(void *data, int **channels);
+int omap_free_alsa_sound_dma(void *data, int **channels);
 
-int omap_start_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_audio_stop_dma(struct audio_stream *s);
+void omap_stop_alsa_sound_dma(struct audio_stream *s);
 
 #endif

sound/arm/omap/omap-alsa-tsc2101-mixer.h

+/*
+ * 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 
+ * 		     Everett Coleman II <gcc80x86@fuzzyneural.net>
+ *
+ * Based on the ideas in omap-aic23.c and sa11xx-uda1341.c
+ * Copyright (C) 2005 Instituto Nokia de Tecnologia - INdT - Manaus Brazil
+ * Copyright (C) 2002 Tomas Kasparek <tomas.kasparek@seznam.cz>
+ *
+ * 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
+ * option) any later version.
+ *
+ * THIS SOFTWARE IS PROVIDED ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN
+ * NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT,
+ * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT
+ * NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF
+ * USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
+ * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ *
+ * You should have received a copy of the  GNU General Public License along
+ * with this program; if not, write  to the Free Software Foundation, Inc.,
+ * 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * History:
+ *
+ * 2006-03-01   Mika Laitio - Mixer for the tsc2101 driver used in omap boards.
+ * 		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 
+ * 		are still missing.
+ */
+
+#ifndef OMAPALSATSC2101MIXER_H_
+#define OMAPALSATSC2101MIXER_H_
+
+#include <asm/hardware/tsc2101.h>
+#include <../drivers/ssi/omap-tsc2101.h>
+#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_RANGE		(OUTPUT_VOLUME_MIN - OUTPUT_VOLUME_MAX)
+
+/* use input vol of 75 for 0dB gain */
+#define INPUT_VOLUME_MIN 		0x0
+#define INPUT_VOLUME_MAX		0x7D
+#define INPUT_VOLUME_RANGE		(INPUT_VOLUME_MAX - INPUT_VOLUME_MIN)
+
+#define PLAYBACK_TARGET_COUNT		0x02
+#define PLAYBACK_TARGET_LOUDSPEAKER	0x00
+#define PLAYBACK_TARGET_HEADPHONE	0x01
+
+/* 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	MPC_MICSEL(0)	// oss code referred to MIXER_LINE
+#define REC_SRC_SINGLE_ENDED_MICIN_HND	MPC_MICSEL(1)	// oss code referred to MIXER_MIC
+#define REC_SRC_SINGLE_ENDED_AUX1	MPC_MICSEL(2)
+#define REC_SRC_SINGLE_ENDED_AUX2	MPC_MICSEL(3)
+#define REC_SRC_MICIN_HED_AND_AUX1	MPC_MICSEL(4)
+#define REC_SRC_MICIN_HED_AND_AUX2	MPC_MICSEL(5)
+#define REC_SRC_MICIN_HND_AND_AUX1	MPC_MICSEL(6)
+#define REC_SRC_MICIN_HND_AND_AUX2	MPC_MICSEL(7)
+
+#define DEFAULT_OUTPUT_VOLUME		90	// default output volume to dac dgc
+#define DEFAULT_INPUT_VOLUME		20	// default record volume
+
+#define TSC2101_AUDIO_CODEC_REGISTERS_PAGE2     (2)
+
+#endif /*OMAPALSATSC2101MIXER_H_*/

sound/arm/omap/omap-alsa-tsc2101.h

+/*
+ * arch/arc/mach-omap1/omap-alsa-tsc2101.h
+ * 
+ * Alsa Driver for TSC2101 codec for OMAP platform boards.
+ *
+ * Based on former omap-aic23.h and tsc2101 OSS 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
+ * 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
+ * option) any later version.
+ */
+
+#ifndef OMAP_ALSA_TSC2101_H_
+#define OMAP_ALSA_TSC2101_H_
+
+#include <linux/types.h>
+
+/* Define to set the tsc as the master w.r.t McBSP */
+#define TSC_MASTER
+
+#define NUMBER_SAMPLE_RATES_SUPPORTED	16
+
+/*
+ * AUDIO related MACROS
+ */
+#ifndef DEFAULT_BITPERSAMPLE
+#define DEFAULT_BITPERSAMPLE		16
+#endif
+
+#define DEFAULT_SAMPLE_RATE		44100
+#define CODEC_CLOCK 			12000000
+#define AUDIO_MCBSP        		OMAP_MCBSP1
+
+#define PAGE2_AUDIO_CODEC_REGISTERS	(2)
+
+struct tsc2101_samplerate_reg_info {
+	u16 sample_rate;
+	u8 divisor;
+	u8 fs_44kHz;	/* if 0 48 khz, if 1 44.1 khz fsref */
+};
+
+/*
+ * Defines codec specific functions pointers that can be used from the 
+ * common omap-alse base driver for all omap codecs. (tsc2101 and aic23)
+ */
+inline void tsc2101_configure(void);
+void tsc2101_set_samplerate(long rate);
+void tsc2101_clock_setup(void);
+int tsc2101_clock_on(void);
+int tsc2101_clock_off(void);
+int tsc2101_get_default_samplerate(void);
+
+#endif /*OMAP_ALSA_TSC2101_H_*/