Linux-2.6.12-rc2v2.6.12-rc2

Initial git repository build. I'm not bothering with the full history,
even though we have it. We can create a separate "historical" git
archive of that later if we want to, and in the meantime it's about
3.2GB when imported into git - space that would just make the early
git days unnecessarily complicated, when we don't have a lot of good
infrastructure for it.

Let it rip!

3 files changed, 999 insertions, 0 deletions

diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig
new file mode 100644
index 00000000000..cdacf4d3a38
--- /dev/null
+++ b/sound/arm/Kconfig
@@ -0,0 +1,18 @@
+# ALSA ARM drivers
+
+menu "ALSA ARM devices"
+	depends on SND!=n && ARM
+
+config SND_SA11XX_UDA1341
+	tristate "SA11xx UDA1341TS driver (iPaq H3600)"
+	depends on ARCH_SA1100 && SND && 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.
+
+endmenu
+
diff --git a/sound/arm/Makefile b/sound/arm/Makefile
new file mode 100644
index 00000000000..d7e7dc0c3cd
--- /dev/null
+++ b/sound/arm/Makefile
@@ -0,0 +1,8 @@
+#
+# Makefile for ALSA
+#
+
+snd-sa11xx-uda1341-objs := sa11xx-uda1341.o
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_SA11XX_UDA1341) += snd-sa11xx-uda1341.o 
diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c
new file mode 100644
index 00000000000..174bc032d1a
--- /dev/null
+++ b/sound/arm/sa11xx-uda1341.c
@@ -0,0 +1,973 @@
+/*
+ *  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 1.21 2005/01/28 19:34:04 tiwai 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 <linux/config.h>
+#include <sound/driver.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.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 = NULL;	/* ID for this card */
+
+module_param(id, charp, 0444);
+MODULE_PARM_DESC(id, "ID string for SA1100/SA1111 + UDA1341TS soundcard.");
+
+typedef 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
+	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) */
+	snd_pcm_substream_t *stream;
+}audio_stream_t;
+
+typedef struct snd_card_sa11xx_uda1341 {
+	snd_card_t *card;
+	struct l3_client *uda1341;
+	snd_pcm_t *pcm;
+	long samplerate;
+	audio_stream_t s[2];	/* playback & capture */
+} sa11xx_uda1341_t;
+
+static struct snd_card_sa11xx_uda1341 *sa11xx_uda1341 = NULL;
+
+static unsigned int rates[] = {
+	8000,  10666, 10985, 14647,
+	16000, 21970, 22050, 24000,
+	29400, 32000, 44100, 48000,
+};
+
+static snd_pcm_hw_constraint_list_t hw_constraints_rates = {
+	.count	= ARRAY_SIZE(rates),
+	.list	= rates,
+	.mask	= 0,
+};
+
+/* }}} */
+
+/* {{{ 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(sa11xx_uda1341_t *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(sa11xx_uda1341_t *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(sa11xx_uda1341_t *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(audio_stream_t *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(audio_stream_t *s)
+{
+	sa1100_free_dma(s->dmach);
+	s->dmach = -1;
+}
+
+#else
+
+static int audio_dma_request(audio_stream_t *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(audio_stream_t *s)
+{
+	sa1100_free_dma((s)->dma_regs);
+	(s)->dma_regs = 0;
+}
+
+#endif
+
+static u_int audio_get_dma_pos(audio_stream_t *s)
+{
+	snd_pcm_substream_t * substream = s->stream;
+	snd_pcm_runtime_t *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(audio_stream_t *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(audio_stream_t *s)
+{
+	snd_pcm_substream_t *substream = s->stream;
+	snd_pcm_runtime_t *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
+{
+	audio_stream_t *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(snd_pcm_substream_t * substream, int cmd)
+{
+	sa11xx_uda1341_t *chip = snd_pcm_substream_chip(substream);
+	int stream_id = substream->pstr->stream;
+	audio_stream_t *s = &chip->s[stream_id];
+	audio_stream_t *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(snd_pcm_substream_t * substream)
+{
+	sa11xx_uda1341_t *chip = snd_pcm_substream_chip(substream);
+	snd_pcm_runtime_t *runtime = substream->runtime;
+	audio_stream_t *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(snd_pcm_substream_t * substream)
+{
+	sa11xx_uda1341_t *chip = snd_pcm_substream_chip(substream);
+	return audio_get_dma_pos(&chip->s[substream->pstr->stream]);
+}
+
+/* }}} */
+
+static snd_pcm_hardware_t 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 snd_pcm_hardware_t 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(snd_pcm_substream_t * substream)
+{
+	sa11xx_uda1341_t *chip = snd_pcm_substream_chip(substream);
+	snd_pcm_runtime_t *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(snd_pcm_substream_t * substream)
+{
+	sa11xx_uda1341_t *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(snd_pcm_substream_t * substream,
+					snd_pcm_hw_params_t * hw_params)
+{
+        
+	return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+}
+
+static int snd_sa11xx_uda1341_hw_free(snd_pcm_substream_t * substream)
+{
+	return snd_pcm_lib_free_pages(substream);
+}
+
+/* }}} */
+
+static snd_pcm_ops_t 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 snd_pcm_ops_t 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(sa11xx_uda1341_t *sa11xx_uda1341, int device)
+{
+	snd_pcm_t *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_ISA, 
+					      snd_pcm_dma_flags(0),
+					      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(snd_card_t *card, pm_message_t state)
+{
+	sa11xx_uda1341_t *chip = card->pm_private_data;
+
+	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(snd_card_t *card)
+{
+	sa11xx_uda1341_t *chip = card->pm_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
+	return 0;
+}
+#endif /* COMFIG_PM */
+
+void snd_sa11xx_uda1341_free(snd_card_t *card)
+{
+	sa11xx_uda1341_t *chip = card->private_data;
+
+	audio_dma_free(&chip->s[SNDRV_PCM_STREAM_PLAYBACK]);
+	audio_dma_free(&chip->s[SNDRV_PCM_STREAM_CAPTURE]);
+	sa11xx_uda1341 = NULL;
+	card->private_data = NULL;
+	kfree(chip);
+}
+
+static int __init sa11xx_uda1341_init(void)
+{
+	int err;
+	snd_card_t *card;
+
+	if (!machine_is_h3xxx())
+		return -ENODEV;
+
+	/* register the soundcard */
+	card = snd_card_new(-1, id, THIS_MODULE, sizeof(sa11xx_uda1341_t));
+	if (card == NULL)
+		return -ENOMEM;
+
+	sa11xx_uda1341 = kcalloc(1, sizeof(*sa11xx_uda1341), GFP_KERNEL);
+	if (sa11xx_uda1341 == NULL)
+		return -ENOMEM;	
+	spin_lock_init(&chip->s[0].dma_lock);
+	spin_lock_init(&chip->s[1].dma_lock);
+         
+	card->private_data = (void *)sa11xx_uda1341;
+	card->private_free = snd_sa11xx_uda1341_free;
+
+	sa11xx_uda1341->card = card;
+	sa11xx_uda1341->samplerate = AUDIO_RATE_DEFAULT;
+
+	// mixer
+	if ((err = snd_chip_uda1341_mixer_new(sa11xx_uda1341->card, &sa11xx_uda1341->uda1341)))
+		goto nodev;
+
+	// PCM
+	if ((err = snd_card_sa11xx_uda1341_pcm(sa11xx_uda1341, 0)) < 0)
+		goto nodev;
+        
+	snd_card_set_generic_pm_callback(card,
+				     snd_sa11xx_uda1341_suspend, snd_sa11_uda1341_resume,
+				     sa11xx_uda1341);
+
+	strcpy(card->driver, "UDA1341");
+	strcpy(card->shortname, "H3600 UDA1341TS");
+	sprintf(card->longname, "Compaq iPAQ H3600 with Philips UDA1341TS");
+        
+	if ((err = snd_card_register(card)) == 0) {
+		printk( KERN_INFO "iPAQ audio support initialized\n" );
+		return 0;
+	}
+        
+ nodev:
+	snd_card_free(card);
+	return err;
+}
+
+static void __exit sa11xx_uda1341_exit(void)
+{
+	snd_card_free(sa11xx_uda1341->card);
+}
+
+module_init(sa11xx_uda1341_init);
+module_exit(sa11xx_uda1341_exit);
+
+/* }}} */
+
+/*
+ * Local variables:
+ * indent-tabs-mode: t
+ * End:
+ */