https://github.com/torvalds/linux
Tip revision: 4a6908a3a050aacc9c3a2f36b276b46c0629ad91 authored by Linus Torvalds on 24 December 2008, 23:26:37 UTC
Linux 2.6.28
Linux 2.6.28
Tip revision: 4a6908a
sa11xx-uda1341.c
/*
* 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 */
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:
*/
