Add the rt linux 4.1.3-rt3 as base

Import the rt linux 4.1.3-rt3 as OPNFV kvm base.

It's from git://git.kernel.org/pub/scm/linux/kernel/git/rt/linux-rt-devel.git linux-4.1.y-rt and
the base is:

commit 0917f823c59692d751951bf5ea699a2d1e2f26a2
Author: Sebastian Andrzej Siewior <bigeasy@linutronix.de>
Date:   Sat Jul 25 12:13:34 2015 +0200

    Prepare v4.1.3-rt3

    Signed-off-by: Sebastian Andrzej Siewior <bigeasy@linutronix.de>

We lose all the git history this way and it's not good. We
should apply another opnfv project repo in future.

Change-Id: I87543d81c9df70d99c5001fbdf646b202c19f423
Signed-off-by: Yunhong Jiang <yunhong.jiang@intel.com>

1 files changed, 738 insertions, 0 deletions

diff --git a/kernel/sound/oss/dmasound/dmasound_paula.c b/kernel/sound/oss/dmasound/dmasound_paula.c
new file mode 100644
index 000000000..3f6536186
--- /dev/null
+++ b/kernel/sound/oss/dmasound/dmasound_paula.c
@@ -0,0 +1,738 @@
+/*
+ *  linux/sound/oss/dmasound/dmasound_paula.c
+ *
+ *  Amiga `Paula' DMA Sound Driver
+ *
+ *  See linux/sound/oss/dmasound/dmasound_core.c for copyright and credits
+ *  prior to 28/01/2001
+ *
+ *  28/01/2001 [0.1] Iain Sandoe
+ *		     - added versioning
+ *		     - put in and populated the hardware_afmts field.
+ *             [0.2] - put in SNDCTL_DSP_GETCAPS value.
+ *	       [0.3] - put in constraint on state buffer usage.
+ *	       [0.4] - put in default hard/soft settings
+*/
+
+
+#include <linux/module.h>
+#include <linux/mm.h>
+#include <linux/init.h>
+#include <linux/ioport.h>
+#include <linux/soundcard.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+
+#include <asm/uaccess.h>
+#include <asm/setup.h>
+#include <asm/amigahw.h>
+#include <asm/amigaints.h>
+#include <asm/machdep.h>
+
+#include "dmasound.h"
+
+#define DMASOUND_PAULA_REVISION 0
+#define DMASOUND_PAULA_EDITION 4
+
+#define custom amiga_custom
+   /*
+    *	The minimum period for audio depends on htotal (for OCS/ECS/AGA)
+    *	(Imported from arch/m68k/amiga/amisound.c)
+    */
+
+extern volatile u_short amiga_audio_min_period;
+
+
+   /*
+    *	amiga_mksound() should be able to restore the period after beeping
+    *	(Imported from arch/m68k/amiga/amisound.c)
+    */
+
+extern u_short amiga_audio_period;
+
+
+   /*
+    *	Audio DMA masks
+    */
+
+#define AMI_AUDIO_OFF	(DMAF_AUD0 | DMAF_AUD1 | DMAF_AUD2 | DMAF_AUD3)
+#define AMI_AUDIO_8	(DMAF_SETCLR | DMAF_MASTER | DMAF_AUD0 | DMAF_AUD1)
+#define AMI_AUDIO_14	(AMI_AUDIO_8 | DMAF_AUD2 | DMAF_AUD3)
+
+
+    /*
+     *  Helper pointers for 16(14)-bit sound
+     */
+
+static int write_sq_block_size_half, write_sq_block_size_quarter;
+
+
+/*** Low level stuff *********************************************************/
+
+
+static void *AmiAlloc(unsigned int size, gfp_t flags);
+static void AmiFree(void *obj, unsigned int size);
+static int AmiIrqInit(void);
+#ifdef MODULE
+static void AmiIrqCleanUp(void);
+#endif
+static void AmiSilence(void);
+static void AmiInit(void);
+static int AmiSetFormat(int format);
+static int AmiSetVolume(int volume);
+static int AmiSetTreble(int treble);
+static void AmiPlayNextFrame(int index);
+static void AmiPlay(void);
+static irqreturn_t AmiInterrupt(int irq, void *dummy);
+
+#ifdef CONFIG_HEARTBEAT
+
+    /*
+     *  Heartbeat interferes with sound since the 7 kHz low-pass filter and the
+     *  power LED are controlled by the same line.
+     */
+
+static void (*saved_heartbeat)(int) = NULL;
+
+static inline void disable_heartbeat(void)
+{
+	if (mach_heartbeat) {
+	    saved_heartbeat = mach_heartbeat;
+	    mach_heartbeat = NULL;
+	}
+	AmiSetTreble(dmasound.treble);
+}
+
+static inline void enable_heartbeat(void)
+{
+	if (saved_heartbeat)
+	    mach_heartbeat = saved_heartbeat;
+}
+#else /* !CONFIG_HEARTBEAT */
+#define disable_heartbeat()	do { } while (0)
+#define enable_heartbeat()	do { } while (0)
+#endif /* !CONFIG_HEARTBEAT */
+
+
+/*** Mid level stuff *********************************************************/
+
+static void AmiMixerInit(void);
+static int AmiMixerIoctl(u_int cmd, u_long arg);
+static int AmiWriteSqSetup(void);
+static int AmiStateInfo(char *buffer, size_t space);
+
+
+/*** Translations ************************************************************/
+
+/* ++TeSche: radically changed for new expanding purposes...
+ *
+ * These two routines now deal with copying/expanding/translating the samples
+ * from user space into our buffer at the right frequency. They take care about
+ * how much data there's actually to read, how much buffer space there is and
+ * to convert samples into the right frequency/encoding. They will only work on
+ * complete samples so it may happen they leave some bytes in the input stream
+ * if the user didn't write a multiple of the current sample size. They both
+ * return the number of bytes they've used from both streams so you may detect
+ * such a situation. Luckily all programs should be able to cope with that.
+ *
+ * I think I've optimized anything as far as one can do in plain C, all
+ * variables should fit in registers and the loops are really short. There's
+ * one loop for every possible situation. Writing a more generalized and thus
+ * parameterized loop would only produce slower code. Feel free to optimize
+ * this in assembler if you like. :)
+ *
+ * I think these routines belong here because they're not yet really hardware
+ * independent, especially the fact that the Falcon can play 16bit samples
+ * only in stereo is hardcoded in both of them!
+ *
+ * ++geert: split in even more functions (one per format)
+ */
+
+
+    /*
+     *  Native format
+     */
+
+static ssize_t ami_ct_s8(const u_char __user *userPtr, size_t userCount,
+			 u_char frame[], ssize_t *frameUsed, ssize_t frameLeft)
+{
+	ssize_t count, used;
+
+	if (!dmasound.soft.stereo) {
+		void *p = &frame[*frameUsed];
+		count = min_t(unsigned long, userCount, frameLeft) & ~1;
+		used = count;
+		if (copy_from_user(p, userPtr, count))
+			return -EFAULT;
+	} else {
+		u_char *left = &frame[*frameUsed>>1];
+		u_char *right = left+write_sq_block_size_half;
+		count = min_t(unsigned long, userCount, frameLeft)>>1 & ~1;
+		used = count*2;
+		while (count > 0) {
+			if (get_user(*left++, userPtr++)
+			    || get_user(*right++, userPtr++))
+				return -EFAULT;
+			count--;
+		}
+	}
+	*frameUsed += used;
+	return used;
+}
+
+
+    /*
+     *  Copy and convert 8 bit data
+     */
+
+#define GENERATE_AMI_CT8(funcname, convsample)				\
+static ssize_t funcname(const u_char __user *userPtr, size_t userCount,	\
+			u_char frame[], ssize_t *frameUsed,		\
+			ssize_t frameLeft)				\
+{									\
+	ssize_t count, used;						\
+									\
+	if (!dmasound.soft.stereo) {					\
+		u_char *p = &frame[*frameUsed];				\
+		count = min_t(size_t, userCount, frameLeft) & ~1;	\
+		used = count;						\
+		while (count > 0) {					\
+			u_char data;					\
+			if (get_user(data, userPtr++))			\
+				return -EFAULT;				\
+			*p++ = convsample(data);			\
+			count--;					\
+		}							\
+	} else {							\
+		u_char *left = &frame[*frameUsed>>1];			\
+		u_char *right = left+write_sq_block_size_half;		\
+		count = min_t(size_t, userCount, frameLeft)>>1 & ~1;	\
+		used = count*2;						\
+		while (count > 0) {					\
+			u_char data;					\
+			if (get_user(data, userPtr++))			\
+				return -EFAULT;				\
+			*left++ = convsample(data);			\
+			if (get_user(data, userPtr++))			\
+				return -EFAULT;				\
+			*right++ = convsample(data);			\
+			count--;					\
+		}							\
+	}								\
+	*frameUsed += used;						\
+	return used;							\
+}
+
+#define AMI_CT_ULAW(x)	(dmasound_ulaw2dma8[(x)])
+#define AMI_CT_ALAW(x)	(dmasound_alaw2dma8[(x)])
+#define AMI_CT_U8(x)	((x) ^ 0x80)
+
+GENERATE_AMI_CT8(ami_ct_ulaw, AMI_CT_ULAW)
+GENERATE_AMI_CT8(ami_ct_alaw, AMI_CT_ALAW)
+GENERATE_AMI_CT8(ami_ct_u8, AMI_CT_U8)
+
+
+    /*
+     *  Copy and convert 16 bit data
+     */
+
+#define GENERATE_AMI_CT_16(funcname, convsample)			\
+static ssize_t funcname(const u_char __user *userPtr, size_t userCount,	\
+			u_char frame[], ssize_t *frameUsed,		\
+			ssize_t frameLeft)				\
+{									\
+	const u_short __user *ptr = (const u_short __user *)userPtr;	\
+	ssize_t count, used;						\
+	u_short data;							\
+									\
+	if (!dmasound.soft.stereo) {					\
+		u_char *high = &frame[*frameUsed>>1];			\
+		u_char *low = high+write_sq_block_size_half;		\
+		count = min_t(size_t, userCount, frameLeft)>>1 & ~1;	\
+		used = count*2;						\
+		while (count > 0) {					\
+			if (get_user(data, ptr++))			\
+				return -EFAULT;				\
+			data = convsample(data);			\
+			*high++ = data>>8;				\
+			*low++ = (data>>2) & 0x3f;			\
+			count--;					\
+		}							\
+	} else {							\
+		u_char *lefth = &frame[*frameUsed>>2];			\
+		u_char *leftl = lefth+write_sq_block_size_quarter;	\
+		u_char *righth = lefth+write_sq_block_size_half;	\
+		u_char *rightl = righth+write_sq_block_size_quarter;	\
+		count = min_t(size_t, userCount, frameLeft)>>2 & ~1;	\
+		used = count*4;						\
+		while (count > 0) {					\
+			if (get_user(data, ptr++))			\
+				return -EFAULT;				\
+			data = convsample(data);			\
+			*lefth++ = data>>8;				\
+			*leftl++ = (data>>2) & 0x3f;			\
+			if (get_user(data, ptr++))			\
+				return -EFAULT;				\
+			data = convsample(data);			\
+			*righth++ = data>>8;				\
+			*rightl++ = (data>>2) & 0x3f;			\
+			count--;					\
+		}							\
+	}								\
+	*frameUsed += used;						\
+	return used;							\
+}
+
+#define AMI_CT_S16BE(x)	(x)
+#define AMI_CT_U16BE(x)	((x) ^ 0x8000)
+#define AMI_CT_S16LE(x)	(le2be16((x)))
+#define AMI_CT_U16LE(x)	(le2be16((x)) ^ 0x8000)
+
+GENERATE_AMI_CT_16(ami_ct_s16be, AMI_CT_S16BE)
+GENERATE_AMI_CT_16(ami_ct_u16be, AMI_CT_U16BE)
+GENERATE_AMI_CT_16(ami_ct_s16le, AMI_CT_S16LE)
+GENERATE_AMI_CT_16(ami_ct_u16le, AMI_CT_U16LE)
+
+
+static TRANS transAmiga = {
+	.ct_ulaw	= ami_ct_ulaw,
+	.ct_alaw	= ami_ct_alaw,
+	.ct_s8		= ami_ct_s8,
+	.ct_u8		= ami_ct_u8,
+	.ct_s16be	= ami_ct_s16be,
+	.ct_u16be	= ami_ct_u16be,
+	.ct_s16le	= ami_ct_s16le,
+	.ct_u16le	= ami_ct_u16le,
+};
+
+/*** Low level stuff *********************************************************/
+
+static inline void StopDMA(void)
+{
+	custom.aud[0].audvol = custom.aud[1].audvol = 0;
+	custom.aud[2].audvol = custom.aud[3].audvol = 0;
+	custom.dmacon = AMI_AUDIO_OFF;
+	enable_heartbeat();
+}
+
+static void *AmiAlloc(unsigned int size, gfp_t flags)
+{
+	return amiga_chip_alloc((long)size, "dmasound [Paula]");
+}
+
+static void AmiFree(void *obj, unsigned int size)
+{
+	amiga_chip_free (obj);
+}
+
+static int __init AmiIrqInit(void)
+{
+	/* turn off DMA for audio channels */
+	StopDMA();
+
+	/* Register interrupt handler. */
+	if (request_irq(IRQ_AMIGA_AUD0, AmiInterrupt, 0, "DMA sound",
+			AmiInterrupt))
+		return 0;
+	return 1;
+}
+
+#ifdef MODULE
+static void AmiIrqCleanUp(void)
+{
+	/* turn off DMA for audio channels */
+	StopDMA();
+	/* release the interrupt */
+	free_irq(IRQ_AMIGA_AUD0, AmiInterrupt);
+}
+#endif /* MODULE */
+
+static void AmiSilence(void)
+{
+	/* turn off DMA for audio channels */
+	StopDMA();
+}
+
+
+static void AmiInit(void)
+{
+	int period, i;
+
+	AmiSilence();
+
+	if (dmasound.soft.speed)
+		period = amiga_colorclock/dmasound.soft.speed-1;
+	else
+		period = amiga_audio_min_period;
+	dmasound.hard = dmasound.soft;
+	dmasound.trans_write = &transAmiga;
+
+	if (period < amiga_audio_min_period) {
+		/* we would need to squeeze the sound, but we won't do that */
+		period = amiga_audio_min_period;
+	} else if (period > 65535) {
+		period = 65535;
+	}
+	dmasound.hard.speed = amiga_colorclock/(period+1);
+
+	for (i = 0; i < 4; i++)
+		custom.aud[i].audper = period;
+	amiga_audio_period = period;
+}
+
+
+static int AmiSetFormat(int format)
+{
+	int size;
+
+	/* Amiga sound DMA supports 8bit and 16bit (pseudo 14 bit) modes */
+
+	switch (format) {
+	case AFMT_QUERY:
+		return dmasound.soft.format;
+	case AFMT_MU_LAW:
+	case AFMT_A_LAW:
+	case AFMT_U8:
+	case AFMT_S8:
+		size = 8;
+		break;
+	case AFMT_S16_BE:
+	case AFMT_U16_BE:
+	case AFMT_S16_LE:
+	case AFMT_U16_LE:
+		size = 16;
+		break;
+	default: /* :-) */
+		size = 8;
+		format = AFMT_S8;
+	}
+
+	dmasound.soft.format = format;
+	dmasound.soft.size = size;
+	if (dmasound.minDev == SND_DEV_DSP) {
+		dmasound.dsp.format = format;
+		dmasound.dsp.size = dmasound.soft.size;
+	}
+	AmiInit();
+
+	return format;
+}
+
+
+#define VOLUME_VOXWARE_TO_AMI(v) \
+	(((v) < 0) ? 0 : ((v) > 100) ? 64 : ((v) * 64)/100)
+#define VOLUME_AMI_TO_VOXWARE(v) ((v)*100/64)
+
+static int AmiSetVolume(int volume)
+{
+	dmasound.volume_left = VOLUME_VOXWARE_TO_AMI(volume & 0xff);
+	custom.aud[0].audvol = dmasound.volume_left;
+	dmasound.volume_right = VOLUME_VOXWARE_TO_AMI((volume & 0xff00) >> 8);
+	custom.aud[1].audvol = dmasound.volume_right;
+	if (dmasound.hard.size == 16) {
+		if (dmasound.volume_left == 64 && dmasound.volume_right == 64) {
+			custom.aud[2].audvol = 1;
+			custom.aud[3].audvol = 1;
+		} else {
+			custom.aud[2].audvol = 0;
+			custom.aud[3].audvol = 0;
+		}
+	}
+	return VOLUME_AMI_TO_VOXWARE(dmasound.volume_left) |
+	       (VOLUME_AMI_TO_VOXWARE(dmasound.volume_right) << 8);
+}
+
+static int AmiSetTreble(int treble)
+{
+	dmasound.treble = treble;
+	if (treble < 50)
+		ciaa.pra &= ~0x02;
+	else
+		ciaa.pra |= 0x02;
+	return treble;
+}
+
+
+#define AMI_PLAY_LOADED		1
+#define AMI_PLAY_PLAYING	2
+#define AMI_PLAY_MASK		3
+
+
+static void AmiPlayNextFrame(int index)
+{
+	u_char *start, *ch0, *ch1, *ch2, *ch3;
+	u_long size;
+
+	/* used by AmiPlay() if all doubts whether there really is something
+	 * to be played are already wiped out.
+	 */
+	start = write_sq.buffers[write_sq.front];
+	size = (write_sq.count == index ? write_sq.rear_size
+					: write_sq.block_size)>>1;
+
+	if (dmasound.hard.stereo) {
+		ch0 = start;
+		ch1 = start+write_sq_block_size_half;
+		size >>= 1;
+	} else {
+		ch0 = start;
+		ch1 = start;
+	}
+
+	disable_heartbeat();
+	custom.aud[0].audvol = dmasound.volume_left;
+	custom.aud[1].audvol = dmasound.volume_right;
+	if (dmasound.hard.size == 8) {
+		custom.aud[0].audlc = (u_short *)ZTWO_PADDR(ch0);
+		custom.aud[0].audlen = size;
+		custom.aud[1].audlc = (u_short *)ZTWO_PADDR(ch1);
+		custom.aud[1].audlen = size;
+		custom.dmacon = AMI_AUDIO_8;
+	} else {
+		size >>= 1;
+		custom.aud[0].audlc = (u_short *)ZTWO_PADDR(ch0);
+		custom.aud[0].audlen = size;
+		custom.aud[1].audlc = (u_short *)ZTWO_PADDR(ch1);
+		custom.aud[1].audlen = size;
+		if (dmasound.volume_left == 64 && dmasound.volume_right == 64) {
+			/* We can play pseudo 14-bit only with the maximum volume */
+			ch3 = ch0+write_sq_block_size_quarter;
+			ch2 = ch1+write_sq_block_size_quarter;
+			custom.aud[2].audvol = 1;  /* we are being affected by the beeps */
+			custom.aud[3].audvol = 1;  /* restoring volume here helps a bit */
+			custom.aud[2].audlc = (u_short *)ZTWO_PADDR(ch2);
+			custom.aud[2].audlen = size;
+			custom.aud[3].audlc = (u_short *)ZTWO_PADDR(ch3);
+			custom.aud[3].audlen = size;
+			custom.dmacon = AMI_AUDIO_14;
+		} else {
+			custom.aud[2].audvol = 0;
+			custom.aud[3].audvol = 0;
+			custom.dmacon = AMI_AUDIO_8;
+		}
+	}
+	write_sq.front = (write_sq.front+1) % write_sq.max_count;
+	write_sq.active |= AMI_PLAY_LOADED;
+}
+
+
+static void AmiPlay(void)
+{
+	int minframes = 1;
+
+	custom.intena = IF_AUD0;
+
+	if (write_sq.active & AMI_PLAY_LOADED) {
+		/* There's already a frame loaded */
+		custom.intena = IF_SETCLR | IF_AUD0;
+		return;
+	}
+
+	if (write_sq.active & AMI_PLAY_PLAYING)
+		/* Increase threshold: frame 1 is already being played */
+		minframes = 2;
+
+	if (write_sq.count < minframes) {
+		/* Nothing to do */
+		custom.intena = IF_SETCLR | IF_AUD0;
+		return;
+	}
+
+	if (write_sq.count <= minframes &&
+	    write_sq.rear_size < write_sq.block_size && !write_sq.syncing) {
+		/* hmmm, the only existing frame is not
+		 * yet filled and we're not syncing?
+		 */
+		custom.intena = IF_SETCLR | IF_AUD0;
+		return;
+	}
+
+	AmiPlayNextFrame(minframes);
+
+	custom.intena = IF_SETCLR | IF_AUD0;
+}
+
+
+static irqreturn_t AmiInterrupt(int irq, void *dummy)
+{
+	int minframes = 1;
+
+	custom.intena = IF_AUD0;
+
+	if (!write_sq.active) {
+		/* Playing was interrupted and sq_reset() has already cleared
+		 * the sq variables, so better don't do anything here.
+		 */
+		WAKE_UP(write_sq.sync_queue);
+		return IRQ_HANDLED;
+	}
+
+	if (write_sq.active & AMI_PLAY_PLAYING) {
+		/* We've just finished a frame */
+		write_sq.count--;
+		WAKE_UP(write_sq.action_queue);
+	}
+
+	if (write_sq.active & AMI_PLAY_LOADED)
+		/* Increase threshold: frame 1 is already being played */
+		minframes = 2;
+
+	/* Shift the flags */
+	write_sq.active = (write_sq.active<<1) & AMI_PLAY_MASK;
+
+	if (!write_sq.active)
+		/* No frame is playing, disable audio DMA */
+		StopDMA();
+
+	custom.intena = IF_SETCLR | IF_AUD0;
+
+	if (write_sq.count >= minframes)
+		/* Try to play the next frame */
+		AmiPlay();
+
+	if (!write_sq.active)
+		/* Nothing to play anymore.
+		   Wake up a process waiting for audio output to drain. */
+		WAKE_UP(write_sq.sync_queue);
+	return IRQ_HANDLED;
+}
+
+/*** Mid level stuff *********************************************************/
+
+
+/*
+ * /dev/mixer abstraction
+ */
+
+static void __init AmiMixerInit(void)
+{
+	dmasound.volume_left = 64;
+	dmasound.volume_right = 64;
+	custom.aud[0].audvol = dmasound.volume_left;
+	custom.aud[3].audvol = 1;	/* For pseudo 14bit */
+	custom.aud[1].audvol = dmasound.volume_right;
+	custom.aud[2].audvol = 1;	/* For pseudo 14bit */
+	dmasound.treble = 50;
+}
+
+static int AmiMixerIoctl(u_int cmd, u_long arg)
+{
+	int data;
+	switch (cmd) {
+	    case SOUND_MIXER_READ_DEVMASK:
+		    return IOCTL_OUT(arg, SOUND_MASK_VOLUME | SOUND_MASK_TREBLE);
+	    case SOUND_MIXER_READ_RECMASK:
+		    return IOCTL_OUT(arg, 0);
+	    case SOUND_MIXER_READ_STEREODEVS:
+		    return IOCTL_OUT(arg, SOUND_MASK_VOLUME);
+	    case SOUND_MIXER_READ_VOLUME:
+		    return IOCTL_OUT(arg,
+			    VOLUME_AMI_TO_VOXWARE(dmasound.volume_left) |
+			    VOLUME_AMI_TO_VOXWARE(dmasound.volume_right) << 8);
+	    case SOUND_MIXER_WRITE_VOLUME:
+		    IOCTL_IN(arg, data);
+		    return IOCTL_OUT(arg, dmasound_set_volume(data));
+	    case SOUND_MIXER_READ_TREBLE:
+		    return IOCTL_OUT(arg, dmasound.treble);
+	    case SOUND_MIXER_WRITE_TREBLE:
+		    IOCTL_IN(arg, data);
+		    return IOCTL_OUT(arg, dmasound_set_treble(data));
+	}
+	return -EINVAL;
+}
+
+
+static int AmiWriteSqSetup(void)
+{
+	write_sq_block_size_half = write_sq.block_size>>1;
+	write_sq_block_size_quarter = write_sq_block_size_half>>1;
+	return 0;
+}
+
+
+static int AmiStateInfo(char *buffer, size_t space)
+{
+	int len = 0;
+	len += sprintf(buffer+len, "\tsound.volume_left = %d [0...64]\n",
+		       dmasound.volume_left);
+	len += sprintf(buffer+len, "\tsound.volume_right = %d [0...64]\n",
+		       dmasound.volume_right);
+	if (len >= space) {
+		printk(KERN_ERR "dmasound_paula: overflowed state buffer alloc.\n") ;
+		len = space ;
+	}
+	return len;
+}
+
+
+/*** Machine definitions *****************************************************/
+
+static SETTINGS def_hard = {
+	.format	= AFMT_S8,
+	.stereo	= 0,
+	.size	= 8,
+	.speed	= 8000
+} ;
+
+static SETTINGS def_soft = {
+	.format	= AFMT_U8,
+	.stereo	= 0,
+	.size	= 8,
+	.speed	= 8000
+} ;
+
+static MACHINE machAmiga = {
+	.name		= "Amiga",
+	.name2		= "AMIGA",
+	.owner		= THIS_MODULE,
+	.dma_alloc	= AmiAlloc,
+	.dma_free	= AmiFree,
+	.irqinit	= AmiIrqInit,
+#ifdef MODULE
+	.irqcleanup	= AmiIrqCleanUp,
+#endif /* MODULE */
+	.init		= AmiInit,
+	.silence	= AmiSilence,
+	.setFormat	= AmiSetFormat,
+	.setVolume	= AmiSetVolume,
+	.setTreble	= AmiSetTreble,
+	.play		= AmiPlay,
+	.mixer_init	= AmiMixerInit,
+	.mixer_ioctl	= AmiMixerIoctl,
+	.write_sq_setup	= AmiWriteSqSetup,
+	.state_info	= AmiStateInfo,
+	.min_dsp_speed	= 8000,
+	.version	= ((DMASOUND_PAULA_REVISION<<8) | DMASOUND_PAULA_EDITION),
+	.hardware_afmts	= (AFMT_S8 | AFMT_S16_BE), /* h'ware-supported formats *only* here */
+	.capabilities	= DSP_CAP_BATCH          /* As per SNDCTL_DSP_GETCAPS */
+};
+
+
+/*** Config & Setup **********************************************************/
+
+
+static int __init amiga_audio_probe(struct platform_device *pdev)
+{
+	dmasound.mach = machAmiga;
+	dmasound.mach.default_hard = def_hard ;
+	dmasound.mach.default_soft = def_soft ;
+	return dmasound_init();
+}
+
+static int __exit amiga_audio_remove(struct platform_device *pdev)
+{
+	dmasound_deinit();
+	return 0;
+}
+
+static struct platform_driver amiga_audio_driver = {
+	.remove = __exit_p(amiga_audio_remove),
+	.driver   = {
+		.name	= "amiga-audio",
+	},
+};
+
+module_platform_driver_probe(amiga_audio_driver, amiga_audio_probe);
+
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:amiga-audio");