Merge branch 'master' of /home/davem/src/GIT/linux-2.6/

Conflicts:
	drivers/net/wimax/i2400m/usb-notif.c

6 files changed, 73 insertions, 1009 deletions

diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig
index f8e6de4..885683a 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 2054de1..5a549ed 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/aaci.c b/sound/arm/aaci.c
index 772901e..7fbd68f 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -995,10 +995,11 @@ static struct aaci * __devinit aaci_init_card(struct amba_device *dev)
 {
 	struct aaci *aaci;
 	struct snd_card *card;
+	int err;
 
-	card = snd_card_new(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1,
-			    THIS_MODULE, sizeof(struct aaci));
-	if (card == NULL)
+	err = snd_card_create(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1,
+			      THIS_MODULE, sizeof(struct aaci), &card);
+	if (err < 0)
 		return NULL;
 
 	card->private_free = aaci_free_card;
diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c
index 35afd0c..2e6355f 100644
--- a/sound/arm/pxa2xx-ac97-lib.c
+++ b/sound/arm/pxa2xx-ac97-lib.c
@@ -31,6 +31,7 @@ static DECLARE_WAIT_QUEUE_HEAD(gsr_wq);
 static volatile long gsr_bits;
 static struct clk *ac97_clk;
 static struct clk *ac97conf_clk;
+static int reset_gpio;
 
 /*
  * Beware PXA27x bugs:
@@ -42,6 +43,45 @@ static struct clk *ac97conf_clk;
  * 1 jiffy timeout if interrupt never comes).
  */
 
+enum {
+	RESETGPIO_FORCE_HIGH,
+	RESETGPIO_FORCE_LOW,
+	RESETGPIO_NORMAL_ALTFUNC
+};
+
+/**
+ * set_resetgpio_mode - computes and sets the AC97_RESET gpio mode on PXA
+ * @mode: chosen action
+ *
+ * As the PXA27x CPUs suffer from a AC97 bug, a manual control of the reset line
+ * must be done to insure proper work of AC97 reset line.  This function
+ * computes the correct gpio_mode for further use by reset functions, and
+ * applied the change through pxa_gpio_mode.
+ */
+static void set_resetgpio_mode(int resetgpio_action)
+{
+	int mode = 0;
+
+	if (reset_gpio)
+		switch (resetgpio_action) {
+		case RESETGPIO_NORMAL_ALTFUNC:
+			if (reset_gpio == 113)
+				mode = 113 | GPIO_OUT | GPIO_DFLT_LOW;
+			if (reset_gpio == 95)
+				mode = 95 | GPIO_ALT_FN_1_OUT;
+			break;
+		case RESETGPIO_FORCE_LOW:
+			mode = reset_gpio | GPIO_OUT | GPIO_DFLT_LOW;
+			break;
+		case RESETGPIO_FORCE_HIGH:
+			mode = reset_gpio | GPIO_OUT | GPIO_DFLT_HIGH;
+			break;
+		};
+
+	if (mode)
+		pxa_gpio_mode(mode);
+}
+
 unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
 {
 	unsigned short val = -1;
@@ -137,10 +177,10 @@ static inline void pxa_ac97_warm_pxa27x(void)
 
 	/* warm reset broken on Bulverde,
 	   so manually keep AC97 reset high */
-	pxa_gpio_mode(113 | GPIO_OUT | GPIO_DFLT_HIGH);
+	set_resetgpio_mode(RESETGPIO_FORCE_HIGH);
 	udelay(10);
 	GCR |= GCR_WARM_RST;
-	pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT);
+	set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC);
 	udelay(500);
 }
 
@@ -308,8 +348,8 @@ int pxa2xx_ac97_hw_resume(void)
 		pxa_gpio_mode(GPIO29_SDATA_IN_AC97_MD);
 	}
 	if (cpu_is_pxa27x()) {
-		/* Use GPIO 113 as AC97 Reset on Bulverde */
-		pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT);
+		/* Use GPIO 113 or 95 as AC97 Reset on Bulverde */
+		set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC);
 	}
 	clk_enable(ac97_clk);
 	return 0;
@@ -320,6 +360,27 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_resume);
 int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev)
 {
 	int ret;
+	struct pxa2xx_ac97_platform_data *pdata = dev->dev.platform_data;
+
+	if (pdata) {
+		switch (pdata->reset_gpio) {
+		case 95:
+		case 113:
+			reset_gpio = pdata->reset_gpio;
+			break;
+		case 0:
+			reset_gpio = 113;
+			break;
+		case -1:
+			break;
+		default:
+			dev_err(&dev->dev, "Invalid reset GPIO %d\n",
+				pdata->reset_gpio);
+		}
+	} else {
+		if (cpu_is_pxa27x())
+			reset_gpio = 113;
+	}
 
 	if (cpu_is_pxa25x() || cpu_is_pxa27x()) {
 		pxa_gpio_mode(GPIO31_SYNC_AC97_MD);
@@ -330,7 +391,7 @@ int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev)
 
 	if (cpu_is_pxa27x()) {
 		/* Use GPIO 113 as AC97 Reset on Bulverde */
-		pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT);
+		set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC);
 		ac97conf_clk = clk_get(&dev->dev, "AC97CONFCLK");
 		if (IS_ERR(ac97conf_clk)) {
 			ret = PTR_ERR(ac97conf_clk);
diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c
index 85cf591..7ed100c 100644
--- a/sound/arm/pxa2xx-ac97.c
+++ b/sound/arm/pxa2xx-ac97.c
@@ -173,10 +173,9 @@ static int __devinit pxa2xx_ac97_probe(struct platform_device *dev)
 	struct snd_ac97_template ac97_template;
 	int ret;
 
-	ret = -ENOMEM;
-	card = snd_card_new(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1,
-			    THIS_MODULE, 0);
-	if (!card)
+	ret = snd_card_create(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1,
+			      THIS_MODULE, 0, &card);
+	if (ret < 0)
 		goto err;
 
 	card->dev = &dev->dev;
diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c
deleted file mode 100644
index 1dcd51d..0000000
--- a/sound/arm/sa11xx-uda1341.c
+++ /dev/null
@@ -1,983 +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 */
-	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:
- */