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Commit 6bc09656 authored by Juha Yrjola's avatar Juha Yrjola
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Merge source.mvista.com:linux-omap

parents b6a3bbbf 7508d984
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arch/arm/mach-omap1/board-h2.c
View file @ 6bc09656
......@@ -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
View file @ 6bc09656
......@@ -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
View file @ 6bc09656
......@@ -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
View file @ 6bc09656
/*
* 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
View file @ 6bc09656
......@@ -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
View file @ 6bc09656
......@@ -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-$(CONFIG_SND) += omap/
sound/arm/omap/Makefile 0 → 100644
View file @ 6bc09656
#
## Makefile for ALSA OMAP
#
#
obj-$(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
View file @ 6bc09656
/*
* 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 0 → 100644
View file @ 6bc09656
/*
* 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 0 → 100644
View file @ 6bc09656
/*
* 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
View file @ 6bc09656
/*
* 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
View file @ 6bc09656
/*
* 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.c 0 → 100644
View file @ 6bc09656
This diff is collapsed.
sound/arm/omap/omap-alsa-tsc2101-mixer.h 0 → 100644
View file @ 6bc09656
/*
* 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.c 0 → 100644
View file @ 6bc09656
This diff is collapsed.
sound/arm/omap/omap-alsa-tsc2101.h 0 → 100644
View file @ 6bc09656
/*
* 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_*/
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