1013 строки
27 KiB
C
1013 строки
27 KiB
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)
|
|
*/
|
|
|
|
/* $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:
|
|
*/
|