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linux/sound/arm/sa11xx-uda1341.c
Yoann Padioleau 632155e659 potential parse error in ifdef 
I have made a tool to parse the kernel that does not pre-process the
source.  That means that my parser tries to parse all the code, including
code in the #else branch or code that is not often compiled because the
driver is not very used (or not used at all).  So, my parser sometimes
reports parse error not originally detected by gcc.  Here is my (first)
patch.

[akpm@linux-foundation.org: fix amd8111e.c]
Signed-off-by: Yoann Padioleau <padator@wanadoo.fr>
Acked-by: Matthew Wilcox <matthew@wil.cx>
Acked-by: Wim Van Sebroeck <wim@iguana.be>
Acked-by: David Woodhouse <dwmw2@infradead.org>
Acked-by: Jeff Garzik <jeff@garzik.org>
Acked-by: James Bottomley <James.Bottomley@steeleye.com>
Cc: Russell King <rmk@arm.linux.org.uk>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
2007-06-01 08:18:27 -07:00

1012 lines
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/*
* 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)
*/
/* $Id: sa11xx-uda1341.c,v 1.27 2005/12/07 09:13:42 cladisch Exp $ */
/***************************************************************************************************
*
* 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 <sound/driver.h>
#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 <asm/hardware.h>
#include <asm/arch/h3600.h>
#include <asm/mach-types.h>
#include <asm/dma.h>
#ifdef CONFIG_H3600_HAL
#include <asm/semaphore.h>
#include <asm/uaccess.h>
#include <asm/arch/h3600_hal.h>
#endif
#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
*/
#ifdef CONFIG_H3600_HAL
#define HH_VERSION 1
#endif
/* {{{ 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 */
#ifdef CONFIG_H3600_HAL
h3600_audio_clock(rate);
#else
sa11xx_uda1341_set_audio_clock(rate);
#endif
/* 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 */
#ifdef CONFIG_H3600_HAL
h3600_audio_power(AUDIO_RATE_DEFAULT);
#else
clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
set_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON);
set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
#endif
/* 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;
#ifdef CONFIG_H3600_HAL
h3600_audio_mute(0);
#else
clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
#endif
}
static void sa11xx_uda1341_audio_shutdown(struct sa11xx_uda1341 *sa11xx_uda1341)
{
/* mute on */
#ifdef CONFIG_H3600_HAL
h3600_audio_mute(1);
#else
set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
#endif
/* 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??? */
#ifdef CONFIG_H3600_HAL
h3600_audio_power(0);
h3600_audio_mute(0);
#else
clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON);
clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
#endif
}
/* }}} */
/* {{{ 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) {
snd_assert(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_assert(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 __init sa11xx_uda1341_probe(struct platform_device *devptr)
{
int err;
struct snd_card *card;
struct sa11xx_uda1341 *chip;
/* register the soundcard */
card = snd_card_new(-1, id, THIS_MODULE, sizeof(struct sa11xx_uda1341));
if (card == NULL)
return -ENOMEM;
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:
*/
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