ALSA: drop outdated and broken sa11xx-uda1341 driver

It depends on L3 support from 2.4 kernel (CONFIG_L3) that never got
merged into mainline. Since there's no way to use it on any of
supported machines (iPaq h3100 or h3600), better drop it for now.
It can be reimplemented later using ASoC infrastructure (there's
already a driver for uda1341 codec in mainline, so only CPU and machine
parts need to be written).

Signed-off-by: Dmitry Artamonow <mad_soft@inbox.ru>
Cc: Russell King <linux@arm.linux.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

3 files changed, 0 insertions, 998 deletions

diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig
index f8e6de48d816..885683a3b0bd 100644
--- a/sound/arm/Kconfig
+++ b/sound/arm/Kconfig
@@ -11,17 +11,6 @@ menuconfig 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
 	tristate "ARM PrimeCell PL041 AC Link support"
 	depends on ARM_AMBA
diff --git a/sound/arm/Makefile b/sound/arm/Makefile
index 2054de11de8a..5a549ed6c8aa 100644
--- a/sound/arm/Makefile
+++ b/sound/arm/Makefile
@@ -2,9 +2,6 @@
 # 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
 snd-aaci-objs			:= aaci.o devdma.o
 
diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c
deleted file mode 100644
index 7101d3d8bae6..000000000000
--- a/sound/arm/sa11xx-uda1341.c
+++ /dev/null
@@ -1,984 +0,0 @@
-/*
- *  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:
- */