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Commit ae02cde7 authored by Takashi Iwai's avatar Takashi Iwai
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Merge branch 'topic/drop-l3' into for-linus

parents f11a936f 323a5961
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include/sound/uda1341.h deleted 100644 → 0
View file @ f11a936f
/*
* linux/include/linux/l3/uda1341.h
*
* Philips UDA1341 mixer device driver for ALSA
*
* 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.
*
* History:
*
* 2002-03-13 Tomas Kasparek Initial release - based on uda1341.h from OSS
* 2002-03-30 Tomas Kasparek Proc filesystem support, complete mixer and DSP
* features support
*/
#define UDA1341_ALSA_NAME "snd-uda1341"
/*
* Default rate set after inicialization
*/
#define AUDIO_RATE_DEFAULT 44100
/*
* UDA1341 L3 address and command types
*/
#define UDA1341_L3ADDR 5
#define UDA1341_DATA0 (UDA1341_L3ADDR << 2 | 0)
#define UDA1341_DATA1 (UDA1341_L3ADDR << 2 | 1)
#define UDA1341_STATUS (UDA1341_L3ADDR << 2 | 2)
enum uda1341_onoff {
OFF=0,
ON,
};
enum uda1341_format {
I2S=0,
LSB16,
LSB18,
LSB20,
MSB,
LSB16MSB,
LSB18MSB,
LSB20MSB,
};
enum uda1341_fs {
F512=0,
F384,
F256,
Funused,
};
enum uda1341_peak {
BEFORE=0,
AFTER,
};
enum uda1341_filter {
FLAT=0,
MIN,
MIN2,
MAX,
};
enum uda1341_mixer {
DOUBLE,
LINE,
MIC,
MIXER,
};
enum uda1341_deemp {
NONE,
D32,
D44,
D48,
};
enum uda1341_config {
CMD_READ_REG = 0,
CMD_RESET,
CMD_FS,
CMD_FORMAT,
CMD_OGAIN,
CMD_IGAIN,
CMD_DAC,
CMD_ADC,
CMD_VOLUME,
CMD_BASS,
CMD_TREBBLE,
CMD_PEAK,
CMD_DEEMP,
CMD_MUTE,
CMD_FILTER,
CMD_CH1,
CMD_CH2,
CMD_MIC,
CMD_MIXER,
CMD_AGC,
CMD_IG,
CMD_AGC_TIME,
CMD_AGC_LEVEL,
#ifdef CONFIG_PM
CMD_SUSPEND,
CMD_RESUME,
#endif
CMD_LAST,
};
enum write_through {
//used in update_bits (write_cfg) to avoid l3_write - just update local copy of regs.
REGS_ONLY=0,
//update local regs and write value to uda1341 - do l3_write
FLUSH,
};
int __init snd_chip_uda1341_mixer_new(struct snd_card *card, struct l3_client **clnt);
/*
* Local variables:
* indent-tabs-mode: t
* End:
*/
sound/arm/Kconfig
View file @ ae02cde7
...@@ -11,17 +11,6 @@ menuconfig SND_ARM ...@@ -11,17 +11,6 @@ menuconfig SND_ARM
if SND_ARM if SND_ARM
config SND_SA11XX_UDA1341
tristate "SA11xx UDA1341TS driver (iPaq H3600)"
depends on ARCH_SA1100 && L3
select SND_PCM
help
Say Y here if you have a Compaq iPaq H3x00 handheld computer
and want to use its Philips UDA 1341 audio chip.
To compile this driver as a module, choose M here: the module
will be called snd-sa11xx-uda1341.
config SND_ARMAACI config SND_ARMAACI
tristate "ARM PrimeCell PL041 AC Link support" tristate "ARM PrimeCell PL041 AC Link support"
depends on ARM_AMBA depends on ARM_AMBA
......
sound/arm/Makefile
View file @ ae02cde7
...@@ -2,9 +2,6 @@ ...@@ -2,9 +2,6 @@
# Makefile for ALSA # Makefile for ALSA
# #
obj-$(CONFIG_SND_SA11XX_UDA1341) += snd-sa11xx-uda1341.o
snd-sa11xx-uda1341-objs := sa11xx-uda1341.o
obj-$(CONFIG_SND_ARMAACI) += snd-aaci.o obj-$(CONFIG_SND_ARMAACI) += snd-aaci.o
snd-aaci-objs := aaci.o devdma.o snd-aaci-objs := aaci.o devdma.o
......
sound/arm/sa11xx-uda1341.c deleted 100644 → 0
View file @ f11a936f
/*
* Driver for Philips UDA1341TS on Compaq iPAQ H3600 soundcard
* 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.
*
* History:
*
* 2002-03-13 Tomas Kasparek initial release - based on h3600-uda1341.c from OSS
* 2002-03-20 Tomas Kasparek playback over ALSA is working
* 2002-03-28 Tomas Kasparek playback over OSS emulation is working
* 2002-03-29 Tomas Kasparek basic capture is working (native ALSA)
* 2002-03-29 Tomas Kasparek capture is working (OSS emulation)
* 2002-04-04 Tomas Kasparek better rates handling (allow non-standard rates)
* 2003-02-14 Brian Avery fixed full duplex mode, other updates
* 2003-02-20 Tomas Kasparek merged updates by Brian (except HAL)
* 2003-04-19 Jaroslav Kysela recoded DMA stuff to follow 2.4.18rmk3-hh24 kernel
* working suspend and resume
* 2003-04-28 Tomas Kasparek updated work by Jaroslav to compile it under 2.5.x again
* merged HAL layer (patches from Brian)
*/
/***************************************************************************************************
*
* To understand what Alsa Drivers should be doing look at "Writing an Alsa Driver" by Takashi Iwai
* available in the Alsa doc section on the website
*
* A few notes to make things clearer. The UDA1341 is hooked up to Serial port 4 on the SA1100.
* We are using SSP mode to talk to the UDA1341. The UDA1341 bit & wordselect clocks are generated
* by this UART. Unfortunately, the clock only runs if the transmit buffer has something in it.
* So, if we are just recording, we feed the transmit DMA stream a bunch of 0x0000 so that the
* transmit buffer is full and the clock keeps going. The zeroes come from FLUSH_BASE_PHYS which
* is a mem loc that always decodes to 0's w/ no off chip access.
*
* Some alsa terminology:
* frame => num_channels * sample_size e.g stereo 16 bit is 2 * 16 = 32 bytes
* period => the least number of bytes that will generate an interrupt e.g. we have a 1024 byte
* buffer and 4 periods in the runtime structure this means we'll get an int every 256
* bytes or 4 times per buffer.
* A number of the sizes are in frames rather than bytes, use frames_to_bytes and
* bytes_to_frames to convert. The easiest way to tell the units is to look at the
* type i.e. runtime-> buffer_size is in frames and its type is snd_pcm_uframes_t
*
* Notes about the pointer fxn:
* The pointer fxn needs to return the offset into the dma buffer in frames.
* Interrupts must be blocked before calling the dma_get_pos fxn to avoid race with interrupts.
*
* Notes about pause/resume
* Implementing this would be complicated so it's skipped. The problem case is:
* A full duplex connection is going, then play is paused. At this point you need to start xmitting
* 0's to keep the record active which means you cant just freeze the dma and resume it later you'd
* need to save off the dma info, and restore it properly on a resume. Yeach!
*
* Notes about transfer methods:
* The async write calls fail. I probably need to implement something else to support them?
*
***************************************************************************************************/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/init.h>
#include <linux/err.h>
#include <linux/platform_device.h>
#include <linux/errno.h>
#include <linux/ioctl.h>
#include <linux/delay.h>
#include <linux/slab.h>
#ifdef CONFIG_PM
#include <linux/pm.h>
#endif
#include <mach/hardware.h>
#include <mach/h3600.h>
#include <asm/mach-types.h>
#include <asm/dma.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/initval.h>
#include <linux/l3/l3.h>
#undef DEBUG_MODE
#undef DEBUG_FUNCTION_NAMES
#include <sound/uda1341.h>
/*
* FIXME: Is this enough as autodetection of 2.4.X-rmkY-hhZ kernels?
* We use DMA stuff from 2.4.18-rmk3-hh24 here to be able to compile this
* module for Familiar 0.6.1
*/
/* {{{ Type definitions */
MODULE_AUTHOR("Tomas Kasparek <tomas.kasparek@seznam.cz>");
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("SA1100/SA1111 + UDA1341TS driver for ALSA");
MODULE_SUPPORTED_DEVICE("{{UDA1341,iPAQ H3600 UDA1341TS}}");
static char *id; /* ID for this card */
module_param(id, charp, 0444);
MODULE_PARM_DESC(id, "ID string for SA1100/SA1111 + UDA1341TS soundcard.");
struct audio_stream {
char *id; /* identification string */
int stream_id; /* numeric identification */
dma_device_t dma_dev; /* device identifier for DMA */
#ifdef HH_VERSION
dmach_t dmach; /* dma channel identification */
#else
dma_regs_t *dma_regs; /* points to our DMA registers */
#endif
unsigned 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 tx_spin; /* are we recoding - flag used to do DMA trans. for sync */
unsigned int old_offset;
spinlock_t dma_lock; /* for locking in DMA operations (see dma-sa1100.c in the kernel) */
struct snd_pcm_substream *stream;
};
struct sa11xx_uda1341 {
struct snd_card *card;
struct l3_client *uda1341;
struct snd_pcm *pcm;
long samplerate;
struct audio_stream s[2]; /* playback & capture */
};
static unsigned int rates[] = {
8000, 10666, 10985, 14647,
16000, 21970, 22050, 24000,
29400, 32000, 44100, 48000,
};
static struct snd_pcm_hw_constraint_list hw_constraints_rates = {
.count = ARRAY_SIZE(rates),
.list = rates,
.mask = 0,
};
static struct platform_device *device;
/* }}} */
/* {{{ Clock and sample rate stuff */
/*
* Stop-gap solution until rest of hh.org HAL stuff is merged.
*/
#define GPIO_H3600_CLK_SET0 GPIO_GPIO (12)
#define GPIO_H3600_CLK_SET1 GPIO_GPIO (13)
#ifdef CONFIG_SA1100_H3XXX
#define clr_sa11xx_uda1341_egpio(x) clr_h3600_egpio(x)
#define set_sa11xx_uda1341_egpio(x) set_h3600_egpio(x)
#else
#error This driver could serve H3x00 handhelds only!
#endif
static void sa11xx_uda1341_set_audio_clock(long val)
{
switch (val) {
case 24000: case 32000: case 48000: /* 00: 12.288 MHz */
GPCR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1;
break;
case 22050: case 29400: case 44100: /* 01: 11.2896 MHz */
GPSR = GPIO_H3600_CLK_SET0;
GPCR = GPIO_H3600_CLK_SET1;
break;
case 8000: case 10666: case 16000: /* 10: 4.096 MHz */
GPCR = GPIO_H3600_CLK_SET0;
GPSR = GPIO_H3600_CLK_SET1;
break;
case 10985: case 14647: case 21970: /* 11: 5.6245 MHz */
GPSR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1;
break;
}
}
static void sa11xx_uda1341_set_samplerate(struct sa11xx_uda1341 *sa11xx_uda1341, long rate)
{
int clk_div = 0;
int clk=0;
/* We don't want to mess with clocks when frames are in flight */
Ser4SSCR0 &= ~SSCR0_SSE;
/* wait for any frame to complete */
udelay(125);
/*
* We have the following clock sources:
* 4.096 MHz, 5.6245 MHz, 11.2896 MHz, 12.288 MHz
* Those can be divided either by 256, 384 or 512.
* This makes up 12 combinations for the following samplerates...
*/
if (rate >= 48000)
rate = 48000;
else if (rate >= 44100)
rate = 44100;
else if (rate >= 32000)
rate = 32000;
else if (rate >= 29400)
rate = 29400;
else if (rate >= 24000)
rate = 24000;
else if (rate >= 22050)
rate = 22050;
else if (rate >= 21970)
rate = 21970;
else if (rate >= 16000)
rate = 16000;
else if (rate >= 14647)
rate = 14647;
else if (rate >= 10985)
rate = 10985;
else if (rate >= 10666)
rate = 10666;
else
rate = 8000;
/* Set the external clock generator */
sa11xx_uda1341_set_audio_clock(rate);
/* Select the clock divisor */
switch (rate) {
case 8000:
case 10985:
case 22050:
case 24000:
clk = F512;
clk_div = SSCR0_SerClkDiv(16);
break;
case 16000:
case 21970:
case 44100:
case 48000:
clk = F256;
clk_div = SSCR0_SerClkDiv(8);
break;
case 10666:
case 14647:
case 29400:
case 32000:
clk = F384;
clk_div = SSCR0_SerClkDiv(12);
break;
}
/* FMT setting should be moved away when other FMTs are added (FIXME) */
l3_command(sa11xx_uda1341->uda1341, CMD_FORMAT, (void *)LSB16);
l3_command(sa11xx_uda1341->uda1341, CMD_FS, (void *)clk);
Ser4SSCR0 = (Ser4SSCR0 & ~0xff00) + clk_div + SSCR0_SSE;
sa11xx_uda1341->samplerate = rate;
}
/* }}} */
/* {{{ HW init and shutdown */
static void sa11xx_uda1341_audio_init(struct sa11xx_uda1341 *sa11xx_uda1341)
{
unsigned long flags;
/* Setup DMA stuff */
sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].id = "UDA1341 out";
sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].stream_id = SNDRV_PCM_STREAM_PLAYBACK;
sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].dma_dev = DMA_Ser4SSPWr;
sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].id = "UDA1341 in";
sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].stream_id = SNDRV_PCM_STREAM_CAPTURE;
sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].dma_dev = DMA_Ser4SSPRd;
/* Initialize the UDA1341 internal state */
/* Setup the uarts */
local_irq_save(flags);
GAFR |= (GPIO_SSP_CLK);
GPDR &= ~(GPIO_SSP_CLK);
Ser4SSCR0 = 0;
Ser4SSCR0 = SSCR0_DataSize(16) + SSCR0_TI + SSCR0_SerClkDiv(8);
Ser4SSCR1 = SSCR1_SClkIactL + SSCR1_SClk1P + SSCR1_ExtClk;
Ser4SSCR0 |= SSCR0_SSE;
local_irq_restore(flags);
/* Enable the audio power */
clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
set_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON);
set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
/* Wait for the UDA1341 to wake up */
mdelay(1); //FIXME - was removed by Perex - Why?
/* Initialize the UDA1341 internal state */
l3_open(sa11xx_uda1341->uda1341);
/* external clock configuration (after l3_open - regs must be initialized */
sa11xx_uda1341_set_samplerate(sa11xx_uda1341, sa11xx_uda1341->samplerate);
/* Wait for the UDA1341 to wake up */
set_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
mdelay(1);
/* make the left and right channels unswapped (flip the WS latch) */
Ser4SSDR = 0;
clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
}
static void sa11xx_uda1341_audio_shutdown(struct sa11xx_uda1341 *sa11xx_uda1341)
{
/* mute on */
set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
/* disable the audio power and all signals leading to the audio chip */
l3_close(sa11xx_uda1341->uda1341);
Ser4SSCR0 = 0;
clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
/* power off and mute off */
/* FIXME - is muting off necesary??? */
clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON);
clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
}
/* }}} */
/* {{{ DMA staff */
/*
* these are the address and sizes used to fill the xmit buffer
* so we can get a clock in record only mode
*/
#define FORCE_CLOCK_ADDR (dma_addr_t)FLUSH_BASE_PHYS
#define FORCE_CLOCK_SIZE 4096 // was 2048
// FIXME Why this value exactly - wrote comment
#define DMA_BUF_SIZE 8176 /* <= MAX_DMA_SIZE from asm/arch-sa1100/dma.h */
#ifdef HH_VERSION
static int audio_dma_request(struct audio_stream *s, void (*callback)(void *, int))
{
int ret;
ret = sa1100_request_dma(&s->dmach, s->id, s->dma_dev);
if (ret < 0) {
printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev);
return ret;
}
sa1100_dma_set_callback(s->dmach, callback);
return 0;
}
static inline void audio_dma_free(struct audio_stream *s)
{
sa1100_free_dma(s->dmach);
s->dmach = -1;
}
#else
static int audio_dma_request(struct audio_stream *s, void (*callback)(void *))
{
int ret;
ret = sa1100_request_dma(s->dma_dev, s->id, callback, s, &s->dma_regs);
if (ret < 0)
printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev);
return ret;
}
static void audio_dma_free(struct audio_stream *s)
{
sa1100_free_dma(s->dma_regs);
s->dma_regs = 0;
}
#endif
static u_int audio_get_dma_pos(struct audio_stream *s)
{
struct snd_pcm_substream *substream = s->stream;
struct snd_pcm_runtime *runtime = substream->runtime;
unsigned int offset;
unsigned long flags;
dma_addr_t addr;
// this must be called w/ interrupts locked out see dma-sa1100.c in the kernel
spin_lock_irqsave(&s->dma_lock, flags);
#ifdef HH_VERSION
sa1100_dma_get_current(s->dmach, NULL, &addr);
#else
addr = sa1100_get_dma_pos((s)->dma_regs);
#endif
offset = addr - runtime->dma_addr;
spin_unlock_irqrestore(&s->dma_lock, flags);
offset = bytes_to_frames(runtime,offset);
if (offset >= runtime->buffer_size)
offset = 0;
return offset;
}
/*
* this stops the dma and clears the dma ptrs
*/
static void audio_stop_dma(struct audio_stream *s)
{
unsigned long flags;
spin_lock_irqsave(&s->dma_lock, flags);
s->active = 0;
s->period = 0;
/* this stops the dma channel and clears the buffer ptrs */
#ifdef HH_VERSION
sa1100_dma_flush_all(s->dmach);
#else
sa1100_clear_dma(s->dma_regs);
#endif
spin_unlock_irqrestore(&s->dma_lock, flags);
}
static void audio_process_dma(struct audio_stream *s)
{
struct snd_pcm_substream *substream = s->stream;
struct snd_pcm_runtime *runtime;
unsigned int dma_size;
unsigned int offset;
int ret;
/* we are requested to process synchronization DMA transfer */
if (s->tx_spin) {
if (snd_BUG_ON(s->stream_id != SNDRV_PCM_STREAM_PLAYBACK))
return;
/* fill the xmit dma buffers and return */
#ifdef HH_VERSION
sa1100_dma_set_spin(s->dmach, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE);
#else
while (1) {
ret = sa1100_start_dma(s->dma_regs, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE);
if (ret)
return;
}
#endif
return;
}
/* must be set here - only valid for running streams, not for forced_clock dma fills */
runtime = substream->runtime;
while (s->active && s->periods < runtime->periods) {
dma_size = frames_to_bytes(runtime, runtime->period_size);
if (s->old_offset) {
/* a little trick, we need resume from old position */
offset = frames_to_bytes(runtime, s->old_offset - 1);
s->old_offset = 0;
s->periods = 0;
s->period = offset / dma_size;
offset %= dma_size;
dma_size = dma_size - offset;
if (!dma_size)
continue; /* special case */
} else {
offset = dma_size * s->period;
snd_BUG_ON(dma_size > DMA_BUF_SIZE);
}
#ifdef HH_VERSION
ret = sa1100_dma_queue_buffer(s->dmach, s, runtime->dma_addr + offset, dma_size);
if (ret)
return; //FIXME
#else
ret = sa1100_start_dma((s)->dma_regs, runtime->dma_addr + offset, dma_size);
if (ret) {
printk(KERN_ERR "audio_process_dma: cannot queue DMA buffer (%i)\n", ret);
return;
}
#endif
s->period++;
s->period %= runtime->periods;
s->periods++;
}
}
#ifdef HH_VERSION
static void audio_dma_callback(void *data, int size)
#else
static void audio_dma_callback(void *data)
#endif
{
struct audio_stream *s = data;
/*
* If we are getting a callback for an active stream then we inform
* the PCM middle layer we've finished a period
*/
if (s->active)
snd_pcm_period_elapsed(s->stream);
spin_lock(&s->dma_lock);
if (!s->tx_spin && s->periods > 0)
s->periods--;
audio_process_dma(s);
spin_unlock(&s->dma_lock);
}
/* }}} */
/* {{{ PCM setting */
/* {{{ trigger & timer */
static int snd_sa11xx_uda1341_trigger(struct snd_pcm_substream *substream, int cmd)
{
struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
int stream_id = substream->pstr->stream;
struct audio_stream *s = &chip->s[stream_id];
struct audio_stream *s1 = &chip->s[stream_id ^ 1];
int err = 0;
/* note local interrupts are already disabled in the midlevel code */
spin_lock(&s->dma_lock);
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
/* now we need to make sure a record only stream has a clock */
if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
/* we need to force fill the xmit DMA with zeros */
s1->tx_spin = 1;
audio_process_dma(s1);
}
/* this case is when you were recording then you turn on a
* playback stream so we stop (also clears it) the dma first,
* clear the sync flag and then we let it turned on
*/
else {
s->tx_spin = 0;
}
/* requested stream startup */
s->active = 1;
audio_process_dma(s);
break;
case SNDRV_PCM_TRIGGER_STOP:
/* requested stream shutdown */
audio_stop_dma(s);
/*
* now we need to make sure a record only stream has a clock
* so if we're stopping a playback with an active capture
* we need to turn the 0 fill dma on for the xmit side
*/
if (stream_id == SNDRV_PCM_STREAM_PLAYBACK && s1->active) {
/* we need to force fill the xmit DMA with zeros */
s->tx_spin = 1;
audio_process_dma(s);
}
/*
* we killed a capture only stream, so we should also kill
* the zero fill transmit
*/
else {
if (s1->tx_spin) {
s1->tx_spin = 0;
audio_stop_dma(s1);
}
}
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
s->active = 0;
#ifdef HH_VERSION
sa1100_dma_stop(s->dmach);
#else
//FIXME - DMA API
#endif
s->old_offset = audio_get_dma_pos(s) + 1;
#ifdef HH_VERSION
sa1100_dma_flush_all(s->dmach);
#else
//FIXME - DMA API
#endif
s->periods = 0;
break;
case SNDRV_PCM_TRIGGER_RESUME:
s->active = 1;
s->tx_spin = 0;
audio_process_dma(s);
if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
s1->tx_spin = 1;
audio_process_dma(s1);
}
break;
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
#ifdef HH_VERSION
sa1100_dma_stop(s->dmach);
#else
//FIXME - DMA API
#endif
s->active = 0;
if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) {
if (s1->active) {
s->tx_spin = 1;
s->old_offset = audio_get_dma_pos(s) + 1;
#ifdef HH_VERSION
sa1100_dma_flush_all(s->dmach);
#else
//FIXME - DMA API
#endif
audio_process_dma(s);
}
} else {
if (s1->tx_spin) {
s1->tx_spin = 0;
#ifdef HH_VERSION
sa1100_dma_flush_all(s1->dmach);
#else
//FIXME - DMA API
#endif
}
}
break;
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
s->active = 1;
if (s->old_offset) {
s->tx_spin = 0;
audio_process_dma(s);
break;
}
if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
s1->tx_spin = 1;
audio_process_dma(s1);
}
#ifdef HH_VERSION
sa1100_dma_resume(s->dmach);
#else
//FIXME - DMA API
#endif
break;
default:
err = -EINVAL;
break;
}
spin_unlock(&s->dma_lock);
return err;
}
static int snd_sa11xx_uda1341_prepare(struct snd_pcm_substream *substream)
{
struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
struct audio_stream *s = &chip->s[substream->pstr->stream];
/* set requested samplerate */
sa11xx_uda1341_set_samplerate(chip, runtime->rate);
/* set requestd format when available */
/* set FMT here !!! FIXME */
s->period = 0;
s->periods = 0;
return 0;
}
static snd_pcm_uframes_t snd_sa11xx_uda1341_pointer(struct snd_pcm_substream *substream)
{
struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
return audio_get_dma_pos(&chip->s[substream->pstr->stream]);
}
/* }}} */
static struct snd_pcm_hardware snd_sa11xx_uda1341_capture =
{
.info = (SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
.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_KNOT),
.rate_min = 8000,
.rate_max = 48000,
.channels_min = 2,
.channels_max = 2,
.buffer_bytes_max = 64*1024,
.period_bytes_min = 64,
.period_bytes_max = DMA_BUF_SIZE,
.periods_min = 2,
.periods_max = 255,
.fifo_size = 0,
};
static struct snd_pcm_hardware snd_sa11xx_uda1341_playback =
{
.info = (SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
.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_KNOT),
.rate_min = 8000,
.rate_max = 48000,
.channels_min = 2,
.channels_max = 2,
.buffer_bytes_max = 64*1024,
.period_bytes_min = 64,
.period_bytes_max = DMA_BUF_SIZE,
.periods_min = 2,
.periods_max = 255,
.fifo_size = 0,
};
static int snd_card_sa11xx_uda1341_open(struct snd_pcm_substream *substream)
{
struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
int stream_id = substream->pstr->stream;
int err;
chip->s[stream_id].stream = substream;
if (stream_id == SNDRV_PCM_STREAM_PLAYBACK)
runtime->hw = snd_sa11xx_uda1341_playback;
else
runtime->hw = snd_sa11xx_uda1341_capture;
if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
return err;
if ((err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates)) < 0)
return err;
return 0;
}
static int snd_card_sa11xx_uda1341_close(struct snd_pcm_substream *substream)
{
struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
chip->s[substream->pstr->stream].stream = NULL;
return 0;
}
/* {{{ HW params & free */
static int snd_sa11xx_uda1341_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_sa11xx_uda1341_hw_free(struct snd_pcm_substream *substream)
{
return snd_pcm_lib_free_pages(substream);
}
/* }}} */
static struct snd_pcm_ops snd_card_sa11xx_uda1341_playback_ops = {
.open = snd_card_sa11xx_uda1341_open,
.close = snd_card_sa11xx_uda1341_close,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = snd_sa11xx_uda1341_hw_params,
.hw_free = snd_sa11xx_uda1341_hw_free,
.prepare = snd_sa11xx_uda1341_prepare,
.trigger = snd_sa11xx_uda1341_trigger,
.pointer = snd_sa11xx_uda1341_pointer,
};
static struct snd_pcm_ops snd_card_sa11xx_uda1341_capture_ops = {
.open = snd_card_sa11xx_uda1341_open,
.close = snd_card_sa11xx_uda1341_close,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = snd_sa11xx_uda1341_hw_params,
.hw_free = snd_sa11xx_uda1341_hw_free,
.prepare = snd_sa11xx_uda1341_prepare,
.trigger = snd_sa11xx_uda1341_trigger,
.pointer = snd_sa11xx_uda1341_pointer,
};
static int __init snd_card_sa11xx_uda1341_pcm(struct sa11xx_uda1341 *sa11xx_uda1341, int device)
{
struct snd_pcm *pcm;
int err;
if ((err = snd_pcm_new(sa11xx_uda1341->card, "UDA1341 PCM", device, 1, 1, &pcm)) < 0)
return err;
/*
* this sets up our initial buffers and sets the dma_type to isa.
* isa works but I'm not sure why (or if) it's the right choice
* this may be too large, trying it for now
*/
snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
snd_dma_isa_data(),
64*1024, 64*1024);
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_card_sa11xx_uda1341_playback_ops);
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_card_sa11xx_uda1341_capture_ops);
pcm->private_data = sa11xx_uda1341;
pcm->info_flags = 0;
strcpy(pcm->name, "UDA1341 PCM");
sa11xx_uda1341_audio_init(sa11xx_uda1341);
/* setup DMA controller */
audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK], audio_dma_callback);
audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE], audio_dma_callback);
sa11xx_uda1341->pcm = pcm;
return 0;
}
/* }}} */
/* {{{ module init & exit */
#ifdef CONFIG_PM
static int snd_sa11xx_uda1341_suspend(struct platform_device *devptr,
pm_message_t state)
{
struct snd_card *card = platform_get_drvdata(devptr);
struct sa11xx_uda1341 *chip = card->private_data;
snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
snd_pcm_suspend_all(chip->pcm);
#ifdef HH_VERSION
sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach);
sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach);
#else
//FIXME
#endif
l3_command(chip->uda1341, CMD_SUSPEND, NULL);
sa11xx_uda1341_audio_shutdown(chip);
return 0;
}
static int snd_sa11xx_uda1341_resume(struct platform_device *devptr)
{
struct snd_card *card = platform_get_drvdata(devptr);
struct sa11xx_uda1341 *chip = card->private_data;
sa11xx_uda1341_audio_init(chip);
l3_command(chip->uda1341, CMD_RESUME, NULL);
#ifdef HH_VERSION
sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach);
sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach);
#else
//FIXME
#endif
snd_power_change_state(card, SNDRV_CTL_POWER_D0);
return 0;
}
#endif /* COMFIG_PM */
void snd_sa11xx_uda1341_free(struct snd_card *card)
{
struct sa11xx_uda1341 *chip = card->private_data;
audio_dma_free(&chip->s[SNDRV_PCM_STREAM_PLAYBACK]);
audio_dma_free(&chip->s[SNDRV_PCM_STREAM_CAPTURE]);
}
static int __devinit sa11xx_uda1341_probe(struct platform_device *devptr)
{
int err;
struct snd_card *card;
struct sa11xx_uda1341 *chip;
/* register the soundcard */
err = snd_card_create(-1, id, THIS_MODULE,
sizeof(struct sa11xx_uda1341), &card);
if (err < 0)
return err;
chip = card->private_data;
spin_lock_init(&chip->s[0].dma_lock);
spin_lock_init(&chip->s[1].dma_lock);
card->private_free = snd_sa11xx_uda1341_free;
chip->card = card;
chip->samplerate = AUDIO_RATE_DEFAULT;
// mixer
if ((err = snd_chip_uda1341_mixer_new(card, &chip->uda1341)))
goto nodev;
// PCM
if ((err = snd_card_sa11xx_uda1341_pcm(chip, 0)) < 0)
goto nodev;
strcpy(card->driver, "UDA1341");
strcpy(card->shortname, "H3600 UDA1341TS");
sprintf(card->longname, "Compaq iPAQ H3600 with Philips UDA1341TS");
snd_card_set_dev(card, &devptr->dev);
if ((err = snd_card_register(card)) == 0) {
printk(KERN_INFO "iPAQ audio support initialized\n");
platform_set_drvdata(devptr, card);
return 0;
}
nodev:
snd_card_free(card);
return err;
}
static int __devexit sa11xx_uda1341_remove(struct platform_device *devptr)
{
snd_card_free(platform_get_drvdata(devptr));
platform_set_drvdata(devptr, NULL);
return 0;
}
#define SA11XX_UDA1341_DRIVER "sa11xx_uda1341"
static struct platform_driver sa11xx_uda1341_driver = {
.probe = sa11xx_uda1341_probe,
.remove = __devexit_p(sa11xx_uda1341_remove),
#ifdef CONFIG_PM
.suspend = snd_sa11xx_uda1341_suspend,
.resume = snd_sa11xx_uda1341_resume,
#endif
.driver = {
.name = SA11XX_UDA1341_DRIVER,
},
};
static int __init sa11xx_uda1341_init(void)
{
int err;
if (!machine_is_h3xxx())
return -ENODEV;
if ((err = platform_driver_register(&sa11xx_uda1341_driver)) < 0)
return err;
device = platform_device_register_simple(SA11XX_UDA1341_DRIVER, -1, NULL, 0);
if (!IS_ERR(device)) {
if (platform_get_drvdata(device))
return 0;
platform_device_unregister(device);
err = -ENODEV;
} else
err = PTR_ERR(device);
platform_driver_unregister(&sa11xx_uda1341_driver);
return err;
}
static void __exit sa11xx_uda1341_exit(void)
{
platform_device_unregister(device);
platform_driver_unregister(&sa11xx_uda1341_driver);
}
module_init(sa11xx_uda1341_init);
module_exit(sa11xx_uda1341_exit);
/* }}} */
/*
* Local variables:
* indent-tabs-mode: t
* End:
*/
sound/i2c/Makefile
View file @ ae02cde7
...@@ -7,8 +7,6 @@ snd-i2c-objs := i2c.o ...@@ -7,8 +7,6 @@ snd-i2c-objs := i2c.o
snd-cs8427-objs := cs8427.o snd-cs8427-objs := cs8427.o
snd-tea6330t-objs := tea6330t.o snd-tea6330t-objs := tea6330t.o
obj-$(CONFIG_L3) += l3/
obj-$(CONFIG_SND) += other/ obj-$(CONFIG_SND) += other/
# Toplevel Module Dependency # Toplevel Module Dependency
......
sound/i2c/l3/Makefile deleted 100644 → 0
View file @ f11a936f
#
# Makefile for ALSA
#
snd-uda1341-objs := uda1341.o
# Module Dependency
obj-$(CONFIG_SND_SA11XX_UDA1341) += snd-uda1341.o
sound/i2c/l3/uda1341.c deleted 100644 → 0
View file @ f11a936f
/*
* Philips UDA1341 mixer device driver
* Copyright (c) 2002 Tomas Kasparek <tomas.kasparek@seznam.cz>
*
* Portions are Copyright (C) 2000 Lernout & Hauspie Speech Products, N.V.
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License.
*
* History:
*
* 2002-03-13 Tomas Kasparek initial release - based on uda1341.c from OSS
* 2002-03-28 Tomas Kasparek basic mixer is working (volume, bass, treble)
* 2002-03-30 Tomas Kasparek proc filesystem support, complete mixer and DSP
* features support
* 2002-04-12 Tomas Kasparek proc interface update, code cleanup
* 2002-05-12 Tomas Kasparek another code cleanup
*/
#include <linux/module.h>
#include <linux/init.h>
#include <linux/types.h>
#include <linux/slab.h>
#include <linux/errno.h>
#include <linux/ioctl.h>
#include <asm/uaccess.h>
#include <sound/core.h>
#include <sound/control.h>
#include <sound/initval.h>
#include <sound/info.h>
#include <linux/l3/l3.h>
#include <sound/uda1341.h>
/* {{{ HW regs definition */
#define STAT0 0x00
#define STAT1 0x80
#define STAT_MASK 0x80
#define DATA0_0 0x00
#define DATA0_1 0x40
#define DATA0_2 0x80
#define DATA_MASK 0xc0
#define IS_DATA0(x) ((x) >= data0_0 && (x) <= data0_2)
#define IS_DATA1(x) ((x) == data1)
#define IS_STATUS(x) ((x) == stat0 || (x) == stat1)
#define IS_EXTEND(x) ((x) >= ext0 && (x) <= ext6)
/* }}} */
static const char *peak_names[] = {
"before",
"after",
};
static const char *filter_names[] = {
"flat",
"min",
"min",
"max",
};
static const char *mixer_names[] = {
"double differential",
"input channel 1 (line in)",
"input channel 2 (microphone)",
"digital mixer",
};
static const char *deemp_names[] = {
"none",
"32 kHz",
"44.1 kHz",
"48 kHz",
};
enum uda1341_regs_names {
stat0,
stat1,
data0_0,
data0_1,
data0_2,
data1,
ext0,
ext1,
ext2,
empty,
ext4,
ext5,
ext6,
uda1341_reg_last,
};
static const char *uda1341_reg_names[] = {
"stat 0 ",
"stat 1 ",
"data 00",
"data 01",
"data 02",
"data 1 ",
"ext 0",
"ext 1",
"ext 2",
"empty",
"ext 4",
"ext 5",
"ext 6",
};
static const int uda1341_enum_items[] = {
0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
2, //peak - before/after
4, //deemp - none/32/44.1/48
0,
4, //filter - flat/min/min/max
0, 0, 0,
4, //mixer - differ/line/mic/mixer
0, 0, 0, 0, 0,
};
static const char ** uda1341_enum_names[] = {
NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL,
peak_names, //peak - before/after
deemp_names, //deemp - none/32/44.1/48
NULL,
filter_names, //filter - flat/min/min/max
NULL, NULL, NULL,
mixer_names, //mixer - differ/line/mic/mixer
NULL, NULL, NULL, NULL, NULL,
};
typedef int uda1341_cfg[CMD_LAST];
struct uda1341 {
int (*write) (struct l3_client *uda1341, unsigned short reg, unsigned short val);
int (*read) (struct l3_client *uda1341, unsigned short reg);
unsigned char regs[uda1341_reg_last];
int active;
spinlock_t reg_lock;
struct snd_card *card;
uda1341_cfg cfg;
#ifdef CONFIG_PM
unsigned char suspend_regs[uda1341_reg_last];
uda1341_cfg suspend_cfg;
#endif
};
/* transfer 8bit integer into string with binary representation */
static void int2str_bin8(uint8_t val, char *buf)
{
const int size = sizeof(val) * 8;
int i;
for (i= 0; i < size; i++){
*(buf++) = (val >> (size - 1)) ? '1' : '0';
val <<= 1;
}
*buf = '\0'; //end the string with zero
}
/* {{{ HW manipulation routines */
static int snd_uda1341_codec_write(struct l3_client *clnt, unsigned short reg, unsigned short val)
{
struct uda1341 *uda = clnt->driver_data;
unsigned char buf[2] = { 0xc0, 0xe0 }; // for EXT addressing
int err = 0;
uda->regs[reg] = val;
if (uda->active) {
if (IS_DATA0(reg)) {
err = l3_write(clnt, UDA1341_DATA0, (const unsigned char *)&val, 1);
} else if (IS_DATA1(reg)) {
err = l3_write(clnt, UDA1341_DATA1, (const unsigned char *)&val, 1);
} else if (IS_STATUS(reg)) {
err = l3_write(clnt, UDA1341_STATUS, (const unsigned char *)&val, 1);
} else if (IS_EXTEND(reg)) {
buf[0] |= (reg - ext0) & 0x7; //EXT address
buf[1] |= val; //EXT data
err = l3_write(clnt, UDA1341_DATA0, (const unsigned char *)buf, 2);
}
} else
printk(KERN_ERR "UDA1341 codec not active!\n");
return err;
}
static int snd_uda1341_codec_read(struct l3_client *clnt, unsigned short reg)
{
unsigned char val;
int err;
err = l3_read(clnt, reg, &val, 1);
if (err == 1)
// use just 6bits - the rest is address of the reg
return val & 63;
return err < 0 ? err : -EIO;
}
static inline int snd_uda1341_valid_reg(struct l3_client *clnt, unsigned short reg)
{
return reg < uda1341_reg_last;
}
static int snd_uda1341_update_bits(struct l3_client *clnt, unsigned short reg,
unsigned short mask, unsigned short shift,
unsigned short value, int flush)
{
int change;
unsigned short old, new;
struct uda1341 *uda = clnt->driver_data;
#if 0
printk(KERN_DEBUG "update_bits: reg: %s mask: %d shift: %d val: %d\n",
uda1341_reg_names[reg], mask, shift, value);
#endif
if (!snd_uda1341_valid_reg(clnt, reg))
return -EINVAL;
spin_lock(&uda->reg_lock);
old = uda->regs[reg];
new = (old & ~(mask << shift)) | (value << shift);
change = old != new;
if (change) {
if (flush) uda->write(clnt, reg, new);
uda->regs[reg] = new;
}
spin_unlock(&uda->reg_lock);
return change;
}
static int snd_uda1341_cfg_write(struct l3_client *clnt, unsigned short what,
unsigned short value, int flush)
{
struct uda1341 *uda = clnt->driver_data;
int ret = 0;
#ifdef CONFIG_PM
int reg;
#endif
#if 0
printk(KERN_DEBUG "cfg_write what: %d value: %d\n", what, value);
#endif
uda->cfg[what] = value;
switch(what) {
case CMD_RESET:
ret = snd_uda1341_update_bits(clnt, data0_2, 1, 2, 1, flush); // MUTE
ret = snd_uda1341_update_bits(clnt, stat0, 1, 6, 1, flush); // RESET
ret = snd_uda1341_update_bits(clnt, stat0, 1, 6, 0, flush); // RESTORE
uda->cfg[CMD_RESET]=0;
break;
case CMD_FS:
ret = snd_uda1341_update_bits(clnt, stat0, 3, 4, value, flush);
break;
case CMD_FORMAT:
ret = snd_uda1341_update_bits(clnt, stat0, 7, 1, value, flush);
break;
case CMD_OGAIN:
ret = snd_uda1341_update_bits(clnt, stat1, 1, 6, value, flush);
break;
case CMD_IGAIN:
ret = snd_uda1341_update_bits(clnt, stat1, 1, 5, value, flush);
break;
case CMD_DAC:
ret = snd_uda1341_update_bits(clnt, stat1, 1, 0, value, flush);
break;
case CMD_ADC:
ret = snd_uda1341_update_bits(clnt, stat1, 1, 1, value, flush);
break;
case CMD_VOLUME:
ret = snd_uda1341_update_bits(clnt, data0_0, 63, 0, value, flush);
break;
case CMD_BASS:
ret = snd_uda1341_update_bits(clnt, data0_1, 15, 2, value, flush);
break;
case CMD_TREBBLE:
ret = snd_uda1341_update_bits(clnt, data0_1, 3, 0, value, flush);
break;
case CMD_PEAK:
ret = snd_uda1341_update_bits(clnt, data0_2, 1, 5, value, flush);
break;
case CMD_DEEMP:
ret = snd_uda1341_update_bits(clnt, data0_2, 3, 3, value, flush);
break;
case CMD_MUTE:
ret = snd_uda1341_update_bits(clnt, data0_2, 1, 2, value, flush);
break;
case CMD_FILTER:
ret = snd_uda1341_update_bits(clnt, data0_2, 3, 0, value, flush);
break;
case CMD_CH1:
ret = snd_uda1341_update_bits(clnt, ext0, 31, 0, value, flush);
break;
case CMD_CH2:
ret = snd_uda1341_update_bits(clnt, ext1, 31, 0, value, flush);
break;
case CMD_MIC:
ret = snd_uda1341_update_bits(clnt, ext2, 7, 2, value, flush);
break;
case CMD_MIXER:
ret = snd_uda1341_update_bits(clnt, ext2, 3, 0, value, flush);
break;
case CMD_AGC:
ret = snd_uda1341_update_bits(clnt, ext4, 1, 4, value, flush);
break;
case CMD_IG:
ret = snd_uda1341_update_bits(clnt, ext4, 3, 0, value & 0x3, flush);
ret = snd_uda1341_update_bits(clnt, ext5, 31, 0, value >> 2, flush);
break;
case CMD_AGC_TIME:
ret = snd_uda1341_update_bits(clnt, ext6, 7, 2, value, flush);
break;
case CMD_AGC_LEVEL:
ret = snd_uda1341_update_bits(clnt, ext6, 3, 0, value, flush);
break;
#ifdef CONFIG_PM
case CMD_SUSPEND:
for (reg = stat0; reg < uda1341_reg_last; reg++)
uda->suspend_regs[reg] = uda->regs[reg];
for (reg = 0; reg < CMD_LAST; reg++)
uda->suspend_cfg[reg] = uda->cfg[reg];
break;
case CMD_RESUME:
for (reg = stat0; reg < uda1341_reg_last; reg++)
snd_uda1341_codec_write(clnt, reg, uda->suspend_regs[reg]);
for (reg = 0; reg < CMD_LAST; reg++)
uda->cfg[reg] = uda->suspend_cfg[reg];
break;
#endif
default:
ret = -EINVAL;
break;
}
if (!uda->active)
printk(KERN_ERR "UDA1341 codec not active!\n");
return ret;
}
/* }}} */
/* {{{ Proc interface */
#ifdef CONFIG_PROC_FS
static const char *format_names[] = {
"I2S-bus",
"LSB 16bits",
"LSB 18bits",
"LSB 20bits",
"MSB",
"in LSB 16bits/out MSB",
"in LSB 18bits/out MSB",
"in LSB 20bits/out MSB",
};
static const char *fs_names[] = {
"512*fs",
"384*fs",
"256*fs",
"Unused - bad value!",
};
static const char* bass_values[][16] = {
{"0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB",
"0 dB", "0 dB", "0 dB", "0 dB", "undefined", }, //flat
{"0 dB", "2 dB", "4 dB", "6 dB", "8 dB", "10 dB", "12 dB", "14 dB", "16 dB", "18 dB", "18 dB",
"18 dB", "18 dB", "18 dB", "18 dB", "undefined",}, // min
{"0 dB", "2 dB", "4 dB", "6 dB", "8 dB", "10 dB", "12 dB", "14 dB", "16 dB", "18 dB", "18 dB",
"18 dB", "18 dB", "18 dB", "18 dB", "undefined",}, // min
{"0 dB", "2 dB", "4 dB", "6 dB", "8 dB", "10 dB", "12 dB", "14 dB", "16 dB", "18 dB", "20 dB",
"22 dB", "24 dB", "24 dB", "24 dB", "undefined",}, // max
};
static const char *mic_sens_value[] = {
"-3 dB", "0 dB", "3 dB", "9 dB", "15 dB", "21 dB", "27 dB", "not used",
};
static const unsigned short AGC_atime[] = {
11, 16, 11, 16, 21, 11, 16, 21,
};
static const unsigned short AGC_dtime[] = {
100, 100, 200, 200, 200, 400, 400, 400,
};
static const char *AGC_level[] = {
"-9.0", "-11.5", "-15.0", "-17.5",
};
static const char *ig_small_value[] = {
"-3.0", "-2.5", "-2.0", "-1.5", "-1.0", "-0.5",
};
/*
* this was computed as peak_value[i] = pow((63-i)*1.42,1.013)
*
* UDA1341 datasheet on page 21: Peak value (dB) = (Peak level - 63.5)*5*log2
* There is an table with these values [level]=value: [3]=-90.31, [7]=-84.29
* [61]=-2.78, [62] = -1.48, [63] = 0.0
* I tried to compute it, but using but even using logarithm with base either 10 or 2
* i was'n able to get values in the table from the formula. So I constructed another
* formula (see above) to interpolate the values as good as possible. If there is some
* mistake, please contact me on tomas.kasparek@seznam.cz. Thanks.
* UDA1341TS datasheet is available at:
* http://www-us9.semiconductors.com/acrobat/datasheets/UDA1341TS_3.pdf
*/
static const char *peak_value[] = {
"-INF dB", "N.A.", "N.A", "90.31 dB", "N.A.", "N.A.", "N.A.", "-84.29 dB",
"-82.65 dB", "-81.13 dB", "-79.61 dB", "-78.09 dB", "-76.57 dB", "-75.05 dB", "-73.53 dB",
"-72.01 dB", "-70.49 dB", "-68.97 dB", "-67.45 dB", "-65.93 dB", "-64.41 dB", "-62.90 dB",
"-61.38 dB", "-59.86 dB", "-58.35 dB", "-56.83 dB", "-55.32 dB", "-53.80 dB", "-52.29 dB",
"-50.78 dB", "-49.26 dB", "-47.75 dB", "-46.24 dB", "-44.73 dB", "-43.22 dB", "-41.71 dB",
"-40.20 dB", "-38.69 dB", "-37.19 dB", "-35.68 dB", "-34.17 dB", "-32.67 dB", "-31.17 dB",
"-29.66 dB", "-28.16 dB", "-26.66 dB", "-25.16 dB", "-23.66 dB", "-22.16 dB", "-20.67 dB",
"-19.17 dB", "-17.68 dB", "-16.19 dB", "-14.70 dB", "-13.21 dB", "-11.72 dB", "-10.24 dB",
"-8.76 dB", "-7.28 dB", "-5.81 dB", "-4.34 dB", "-2.88 dB", "-1.43 dB", "0.00 dB",
};
static void snd_uda1341_proc_read(struct snd_info_entry *entry,
struct snd_info_buffer *buffer)
{
struct l3_client *clnt = entry->private_data;
struct uda1341 *uda = clnt->driver_data;
int peak;
peak = snd_uda1341_codec_read(clnt, UDA1341_DATA1);
if (peak < 0)
peak = 0;
snd_iprintf(buffer, "%s\n\n", uda->card->longname);
// for information about computed values see UDA1341TS datasheet pages 15 - 21
snd_iprintf(buffer, "DAC power : %s\n", uda->cfg[CMD_DAC] ? "on" : "off");
snd_iprintf(buffer, "ADC power : %s\n", uda->cfg[CMD_ADC] ? "on" : "off");
snd_iprintf(buffer, "Clock frequency : %s\n", fs_names[uda->cfg[CMD_FS]]);
snd_iprintf(buffer, "Data format : %s\n\n", format_names[uda->cfg[CMD_FORMAT]]);
snd_iprintf(buffer, "Filter mode : %s\n", filter_names[uda->cfg[CMD_FILTER]]);
snd_iprintf(buffer, "Mixer mode : %s\n", mixer_names[uda->cfg[CMD_MIXER]]);
snd_iprintf(buffer, "De-emphasis : %s\n", deemp_names[uda->cfg[CMD_DEEMP]]);
snd_iprintf(buffer, "Peak detection pos. : %s\n", uda->cfg[CMD_PEAK] ? "after" : "before");
snd_iprintf(buffer, "Peak value : %s\n\n", peak_value[peak]);
snd_iprintf(buffer, "Automatic Gain Ctrl : %s\n", uda->cfg[CMD_AGC] ? "on" : "off");
snd_iprintf(buffer, "AGC attack time : %d ms\n", AGC_atime[uda->cfg[CMD_AGC_TIME]]);
snd_iprintf(buffer, "AGC decay time : %d ms\n", AGC_dtime[uda->cfg[CMD_AGC_TIME]]);
snd_iprintf(buffer, "AGC output level : %s dB\n\n", AGC_level[uda->cfg[CMD_AGC_LEVEL]]);
snd_iprintf(buffer, "Mute : %s\n", uda->cfg[CMD_MUTE] ? "on" : "off");
if (uda->cfg[CMD_VOLUME] == 0)
snd_iprintf(buffer, "Volume : 0 dB\n");
else if (uda->cfg[CMD_VOLUME] < 62)
snd_iprintf(buffer, "Volume : %d dB\n", -1*uda->cfg[CMD_VOLUME] +1);
else
snd_iprintf(buffer, "Volume : -INF dB\n");
snd_iprintf(buffer, "Bass : %s\n", bass_values[uda->cfg[CMD_FILTER]][uda->cfg[CMD_BASS]]);
snd_iprintf(buffer, "Trebble : %d dB\n", uda->cfg[CMD_FILTER] ? 2*uda->cfg[CMD_TREBBLE] : 0);
snd_iprintf(buffer, "Input Gain (6dB) : %s\n", uda->cfg[CMD_IGAIN] ? "on" : "off");
snd_iprintf(buffer, "Output Gain (6dB) : %s\n", uda->cfg[CMD_OGAIN] ? "on" : "off");
snd_iprintf(buffer, "Mic sensitivity : %s\n", mic_sens_value[uda->cfg[CMD_MIC]]);
if(uda->cfg[CMD_CH1] < 31)
snd_iprintf(buffer, "Mixer gain channel 1: -%d.%c dB\n",
((uda->cfg[CMD_CH1] >> 1) * 3) + (uda->cfg[CMD_CH1] & 1),
uda->cfg[CMD_CH1] & 1 ? '5' : '0');
else
snd_iprintf(buffer, "Mixer gain channel 1: -INF dB\n");
if(uda->cfg[CMD_CH2] < 31)
snd_iprintf(buffer, "Mixer gain channel 2: -%d.%c dB\n",
((uda->cfg[CMD_CH2] >> 1) * 3) + (uda->cfg[CMD_CH2] & 1),
uda->cfg[CMD_CH2] & 1 ? '5' : '0');
else
snd_iprintf(buffer, "Mixer gain channel 2: -INF dB\n");
if(uda->cfg[CMD_IG] > 5)
snd_iprintf(buffer, "Input Amp. Gain ch 2: %d.%c dB\n",
(uda->cfg[CMD_IG] >> 1) -3, uda->cfg[CMD_IG] & 1 ? '5' : '0');
else
snd_iprintf(buffer, "Input Amp. Gain ch 2: %s dB\n", ig_small_value[uda->cfg[CMD_IG]]);
}
static void snd_uda1341_proc_regs_read(struct snd_info_entry *entry,
struct snd_info_buffer *buffer)
{
struct l3_client *clnt = entry->private_data;
struct uda1341 *uda = clnt->driver_data;
int reg;
char buf[12];
for (reg = 0; reg < uda1341_reg_last; reg ++) {
if (reg == empty)
continue;
int2str_bin8(uda->regs[reg], buf);
snd_iprintf(buffer, "%s = %s\n", uda1341_reg_names[reg], buf);
}
int2str_bin8(snd_uda1341_codec_read(clnt, UDA1341_DATA1), buf);
snd_iprintf(buffer, "DATA1 = %s\n", buf);
}
#endif /* CONFIG_PROC_FS */
static void __devinit snd_uda1341_proc_init(struct snd_card *card, struct l3_client *clnt)
{
struct snd_info_entry *entry;
if (! snd_card_proc_new(card, "uda1341", &entry))
snd_info_set_text_ops(entry, clnt, snd_uda1341_proc_read);
if (! snd_card_proc_new(card, "uda1341-regs", &entry))
snd_info_set_text_ops(entry, clnt, snd_uda1341_proc_regs_read);
}
/* }}} */
/* {{{ Mixer controls setting */
/* {{{ UDA1341 single functions */
#define UDA1341_SINGLE(xname, where, reg, shift, mask, invert) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .info = snd_uda1341_info_single, \
.get = snd_uda1341_get_single, .put = snd_uda1341_put_single, \
.private_value = where | (reg << 5) | (shift << 9) | (mask << 12) | (invert << 18) \
}
static int snd_uda1341_info_single(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
int mask = (kcontrol->private_value >> 12) & 63;
uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = 1;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = mask;
return 0;
}
static int snd_uda1341_get_single(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct l3_client *clnt = snd_kcontrol_chip(kcontrol);
struct uda1341 *uda = clnt->driver_data;
int where = kcontrol->private_value & 31;
int mask = (kcontrol->private_value >> 12) & 63;
int invert = (kcontrol->private_value >> 18) & 1;
ucontrol->value.integer.value[0] = uda->cfg[where];
if (invert)
ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0];
return 0;
}
static int snd_uda1341_put_single(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct l3_client *clnt = snd_kcontrol_chip(kcontrol);
struct uda1341 *uda = clnt->driver_data;
int where = kcontrol->private_value & 31;
int reg = (kcontrol->private_value >> 5) & 15;
int shift = (kcontrol->private_value >> 9) & 7;
int mask = (kcontrol->private_value >> 12) & 63;
int invert = (kcontrol->private_value >> 18) & 1;
unsigned short val;
val = (ucontrol->value.integer.value[0] & mask);
if (invert)
val = mask - val;
uda->cfg[where] = val;
return snd_uda1341_update_bits(clnt, reg, mask, shift, val, FLUSH);
}
/* }}} */
/* {{{ UDA1341 enum functions */
#define UDA1341_ENUM(xname, where, reg, shift, mask, invert) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .info = snd_uda1341_info_enum, \
.get = snd_uda1341_get_enum, .put = snd_uda1341_put_enum, \
.private_value = where | (reg << 5) | (shift << 9) | (mask << 12) | (invert << 18) \
}
static int snd_uda1341_info_enum(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
int where = kcontrol->private_value & 31;
const char **texts;
// this register we don't handle this way
if (!uda1341_enum_items[where])
return -EINVAL;
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
uinfo->value.enumerated.items = uda1341_enum_items[where];
if (uinfo->value.enumerated.item >= uda1341_enum_items[where])
uinfo->value.enumerated.item = uda1341_enum_items[where] - 1;
texts = uda1341_enum_names[where];
strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
return 0;
}
static int snd_uda1341_get_enum(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct l3_client *clnt = snd_kcontrol_chip(kcontrol);
struct uda1341 *uda = clnt->driver_data;
int where = kcontrol->private_value & 31;
ucontrol->value.enumerated.item[0] = uda->cfg[where];
return 0;
}
static int snd_uda1341_put_enum(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct l3_client *clnt = snd_kcontrol_chip(kcontrol);
struct uda1341 *uda = clnt->driver_data;
int where = kcontrol->private_value & 31;
int reg = (kcontrol->private_value >> 5) & 15;
int shift = (kcontrol->private_value >> 9) & 7;
int mask = (kcontrol->private_value >> 12) & 63;
uda->cfg[where] = (ucontrol->value.enumerated.item[0] & mask);
return snd_uda1341_update_bits(clnt, reg, mask, shift, uda->cfg[where], FLUSH);
}
/* }}} */
/* {{{ UDA1341 2regs functions */
#define UDA1341_2REGS(xname, where, reg_1, reg_2, shift_1, shift_2, mask_1, mask_2, invert) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), .info = snd_uda1341_info_2regs, \
.get = snd_uda1341_get_2regs, .put = snd_uda1341_put_2regs, \
.private_value = where | (reg_1 << 5) | (reg_2 << 9) | (shift_1 << 13) | (shift_2 << 16) | \
(mask_1 << 19) | (mask_2 << 25) | (invert << 31) \
}
static int snd_uda1341_info_2regs(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
int mask_1 = (kcontrol->private_value >> 19) & 63;
int mask_2 = (kcontrol->private_value >> 25) & 63;
int mask;
mask = (mask_2 + 1) * (mask_1 + 1) - 1;
uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = 1;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = mask;
return 0;
}
static int snd_uda1341_get_2regs(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct l3_client *clnt = snd_kcontrol_chip(kcontrol);
struct uda1341 *uda = clnt->driver_data;
int where = kcontrol->private_value & 31;
int mask_1 = (kcontrol->private_value >> 19) & 63;
int mask_2 = (kcontrol->private_value >> 25) & 63;
int invert = (kcontrol->private_value >> 31) & 1;
int mask;
mask = (mask_2 + 1) * (mask_1 + 1) - 1;
ucontrol->value.integer.value[0] = uda->cfg[where];
if (invert)
ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0];
return 0;
}
static int snd_uda1341_put_2regs(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct l3_client *clnt = snd_kcontrol_chip(kcontrol);
struct uda1341 *uda = clnt->driver_data;
int where = kcontrol->private_value & 31;
int reg_1 = (kcontrol->private_value >> 5) & 15;
int reg_2 = (kcontrol->private_value >> 9) & 15;
int shift_1 = (kcontrol->private_value >> 13) & 7;
int shift_2 = (kcontrol->private_value >> 16) & 7;
int mask_1 = (kcontrol->private_value >> 19) & 63;
int mask_2 = (kcontrol->private_value >> 25) & 63;
int invert = (kcontrol->private_value >> 31) & 1;
int mask;
unsigned short val1, val2, val;
val = ucontrol->value.integer.value[0];
mask = (mask_2 + 1) * (mask_1 + 1) - 1;
val1 = val & mask_1;
val2 = (val / (mask_1 + 1)) & mask_2;
if (invert) {
val1 = mask_1 - val1;
val2 = mask_2 - val2;
}
uda->cfg[where] = invert ? mask - val : val;
//FIXME - return value
snd_uda1341_update_bits(clnt, reg_1, mask_1, shift_1, val1, FLUSH);
return snd_uda1341_update_bits(clnt, reg_2, mask_2, shift_2, val2, FLUSH);
}
/* }}} */
static struct snd_kcontrol_new snd_uda1341_controls[] = {
UDA1341_SINGLE("Master Playback Switch", CMD_MUTE, data0_2, 2, 1, 1),
UDA1341_SINGLE("Master Playback Volume", CMD_VOLUME, data0_0, 0, 63, 1),
UDA1341_SINGLE("Bass Playback Volume", CMD_BASS, data0_1, 2, 15, 0),
UDA1341_SINGLE("Treble Playback Volume", CMD_TREBBLE, data0_1, 0, 3, 0),
UDA1341_SINGLE("Input Gain Switch", CMD_IGAIN, stat1, 5, 1, 0),
UDA1341_SINGLE("Output Gain Switch", CMD_OGAIN, stat1, 6, 1, 0),
UDA1341_SINGLE("Mixer Gain Channel 1 Volume", CMD_CH1, ext0, 0, 31, 1),
UDA1341_SINGLE("Mixer Gain Channel 2 Volume", CMD_CH2, ext1, 0, 31, 1),
UDA1341_SINGLE("Mic Sensitivity Volume", CMD_MIC, ext2, 2, 7, 0),
UDA1341_SINGLE("AGC Output Level", CMD_AGC_LEVEL, ext6, 0, 3, 0),
UDA1341_SINGLE("AGC Time Constant", CMD_AGC_TIME, ext6, 2, 7, 0),
UDA1341_SINGLE("AGC Time Constant Switch", CMD_AGC, ext4, 4, 1, 0),
UDA1341_SINGLE("DAC Power", CMD_DAC, stat1, 0, 1, 0),
UDA1341_SINGLE("ADC Power", CMD_ADC, stat1, 1, 1, 0),
UDA1341_ENUM("Peak detection", CMD_PEAK, data0_2, 5, 1, 0),
UDA1341_ENUM("De-emphasis", CMD_DEEMP, data0_2, 3, 3, 0),
UDA1341_ENUM("Mixer mode", CMD_MIXER, ext2, 0, 3, 0),
UDA1341_ENUM("Filter mode", CMD_FILTER, data0_2, 0, 3, 0),
UDA1341_2REGS("Gain Input Amplifier Gain (channel 2)", CMD_IG, ext4, ext5, 0, 0, 3, 31, 0),
};
static void uda1341_free(struct l3_client *clnt)
{
l3_detach_client(clnt); // calls kfree for driver_data (struct uda1341)
kfree(clnt);
}
static int uda1341_dev_free(struct snd_device *device)
{
struct l3_client *clnt = device->device_data;
uda1341_free(clnt);
return 0;
}
int __init snd_chip_uda1341_mixer_new(struct snd_card *card, struct l3_client **clntp)
{
static struct snd_device_ops ops = {
.dev_free = uda1341_dev_free,
};
struct l3_client *clnt;
int idx, err;
if (snd_BUG_ON(!card))
return -EINVAL;
clnt = kzalloc(sizeof(*clnt), GFP_KERNEL);
if (clnt == NULL)
return -ENOMEM;
if ((err = l3_attach_client(clnt, "l3-bit-sa1100-gpio", UDA1341_ALSA_NAME))) {
kfree(clnt);
return err;
}
for (idx = 0; idx < ARRAY_SIZE(snd_uda1341_controls); idx++) {
if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_uda1341_controls[idx], clnt))) < 0) {
uda1341_free(clnt);
return err;
}
}
if ((err = snd_device_new(card, SNDRV_DEV_CODEC, clnt, &ops)) < 0) {
uda1341_free(clnt);
return err;
}
*clntp = clnt;
strcpy(card->mixername, "UDA1341TS Mixer");
((struct uda1341 *)clnt->driver_data)->card = card;
snd_uda1341_proc_init(card, clnt);
return 0;
}
/* }}} */
/* {{{ L3 operations */
static int uda1341_attach(struct l3_client *clnt)
{
struct uda1341 *uda;
uda = kzalloc(sizeof(*uda), 0, GFP_KERNEL);
if (!uda)
return -ENOMEM;
/* init fixed parts of my copy of registers */
uda->regs[stat0] = STAT0;
uda->regs[stat1] = STAT1;
uda->regs[data0_0] = DATA0_0;
uda->regs[data0_1] = DATA0_1;
uda->regs[data0_2] = DATA0_2;
uda->write = snd_uda1341_codec_write;
uda->read = snd_uda1341_codec_read;
spin_lock_init(&uda->reg_lock);
clnt->driver_data = uda;
return 0;
}
static void uda1341_detach(struct l3_client *clnt)
{
kfree(clnt->driver_data);
}
static int
uda1341_command(struct l3_client *clnt, int cmd, void *arg)
{
if (cmd != CMD_READ_REG)
return snd_uda1341_cfg_write(clnt, cmd, (int) arg, FLUSH);
return snd_uda1341_codec_read(clnt, (int) arg);
}
static int uda1341_open(struct l3_client *clnt)
{
struct uda1341 *uda = clnt->driver_data;
uda->active = 1;
/* init default configuration */
snd_uda1341_cfg_write(clnt, CMD_RESET, 0, REGS_ONLY);
snd_uda1341_cfg_write(clnt, CMD_FS, F256, FLUSH); // unknown state after reset
snd_uda1341_cfg_write(clnt, CMD_FORMAT, LSB16, FLUSH); // unknown state after reset
snd_uda1341_cfg_write(clnt, CMD_OGAIN, ON, FLUSH); // default off after reset
snd_uda1341_cfg_write(clnt, CMD_IGAIN, ON, FLUSH); // default off after reset
snd_uda1341_cfg_write(clnt, CMD_DAC, ON, FLUSH); // ??? default value after reset
snd_uda1341_cfg_write(clnt, CMD_ADC, ON, FLUSH); // ??? default value after reset
snd_uda1341_cfg_write(clnt, CMD_VOLUME, 20, FLUSH); // default 0dB after reset
snd_uda1341_cfg_write(clnt, CMD_BASS, 0, REGS_ONLY); // default value after reset
snd_uda1341_cfg_write(clnt, CMD_TREBBLE, 0, REGS_ONLY); // default value after reset
snd_uda1341_cfg_write(clnt, CMD_PEAK, AFTER, REGS_ONLY);// default value after reset
snd_uda1341_cfg_write(clnt, CMD_DEEMP, NONE, REGS_ONLY);// default value after reset
//at this moment should be QMUTED by h3600_audio_init
snd_uda1341_cfg_write(clnt, CMD_MUTE, OFF, REGS_ONLY); // default value after reset
snd_uda1341_cfg_write(clnt, CMD_FILTER, MAX, FLUSH); // defaul flat after reset
snd_uda1341_cfg_write(clnt, CMD_CH1, 31, FLUSH); // default value after reset
snd_uda1341_cfg_write(clnt, CMD_CH2, 4, FLUSH); // default value after reset
snd_uda1341_cfg_write(clnt, CMD_MIC, 4, FLUSH); // default 0dB after reset
snd_uda1341_cfg_write(clnt, CMD_MIXER, MIXER, FLUSH); // default doub.dif.mode
snd_uda1341_cfg_write(clnt, CMD_AGC, OFF, FLUSH); // default value after reset
snd_uda1341_cfg_write(clnt, CMD_IG, 0, FLUSH); // unknown state after reset
snd_uda1341_cfg_write(clnt, CMD_AGC_TIME, 0, FLUSH); // default value after reset
snd_uda1341_cfg_write(clnt, CMD_AGC_LEVEL, 0, FLUSH); // default value after reset
return 0;
}
static void uda1341_close(struct l3_client *clnt)
{
struct uda1341 *uda = clnt->driver_data;
uda->active = 0;
}
/* }}} */
/* {{{ Module and L3 initialization */
static struct l3_ops uda1341_ops = {
.open = uda1341_open,
.command = uda1341_command,
.close = uda1341_close,
};
static struct l3_driver uda1341_driver = {
.name = UDA1341_ALSA_NAME,
.attach_client = uda1341_attach,
.detach_client = uda1341_detach,
.ops = &uda1341_ops,
.owner = THIS_MODULE,
};
static int __init uda1341_init(void)
{
return l3_add_driver(&uda1341_driver);
}
static void __exit uda1341_exit(void)
{
l3_del_driver(&uda1341_driver);
}
module_init(uda1341_init);
module_exit(uda1341_exit);
MODULE_AUTHOR("Tomas Kasparek <tomas.kasparek@seznam.cz>");
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Philips UDA1341 CODEC driver for ALSA");
MODULE_SUPPORTED_DEVICE("{{UDA1341,UDA1341TS}}");
EXPORT_SYMBOL(snd_chip_uda1341_mixer_new);
/* }}} */
/*
* Local variables:
* indent-tabs-mode: t
* End:
*/
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