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 000000000000..cdacf4d3a387
--- /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 000000000000..d7e7dc0c3cdf
--- /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 000000000000..174bc032d1ad
--- /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:
+ */