Fixed MTP to work with TWRP

sound/oss/dmasound/Kconfig

@@ -0,0 +1,45 @@
+config DMASOUND_ATARI
+	tristate "Atari DMA sound support"
+	depends on ATARI && SOUND
+	select DMASOUND
+	help
+	  If you want to use the internal audio of your Atari in Linux, answer
+	  Y to this question. This will provide a Sun-like /dev/audio,
+	  compatible with the Linux/i386 sound system. Otherwise, say N.
+
+	  This driver is also available as a module ( = code which can be
+	  inserted in and removed from the running kernel whenever you
+	  want). If you want to compile it as a module, say M here and read
+	  <file:Documentation/kbuild/modules.txt>.
+
+config DMASOUND_PAULA
+	tristate "Amiga DMA sound support"
+	depends on AMIGA && SOUND
+	select DMASOUND
+	help
+	  If you want to use the internal audio of your Amiga in Linux, answer
+	  Y to this question. This will provide a Sun-like /dev/audio,
+	  compatible with the Linux/i386 sound system. Otherwise, say N.
+
+	  This driver is also available as a module ( = code which can be
+	  inserted in and removed from the running kernel whenever you
+	  want). If you want to compile it as a module, say M here and read
+	  <file:Documentation/kbuild/modules.txt>.
+
+config DMASOUND_Q40
+	tristate "Q40 sound support"
+	depends on Q40 && SOUND
+	select DMASOUND
+	help
+	  If you want to use the internal audio of your Q40 in Linux, answer
+	  Y to this question. This will provide a Sun-like /dev/audio,
+	  compatible with the Linux/i386 sound system. Otherwise, say N.
+
+	  This driver is also available as a module ( = code which can be
+	  inserted in and removed from the running kernel whenever you
+	  want). If you want to compile it as a module, say M here and read
+	  <file:Documentation/kbuild/modules.txt>.
+
+config DMASOUND
+	tristate
+	select SOUND_OSS_CORE

sound/oss/dmasound/Makefile

@@ -0,0 +1,7 @@
+#
+# Makefile for the DMA sound driver
+#
+
+obj-$(CONFIG_DMASOUND_ATARI)	+= dmasound_core.o dmasound_atari.o
+obj-$(CONFIG_DMASOUND_PAULA)	+= dmasound_core.o dmasound_paula.o
+obj-$(CONFIG_DMASOUND_Q40)	+= dmasound_core.o dmasound_q40.o

sound/oss/dmasound/dmasound.h

@@ -0,0 +1,261 @@
+#ifndef _dmasound_h_
+/*
+ *  linux/sound/oss/dmasound/dmasound.h
+ *
+ *
+ *  Minor numbers for the sound driver.
+ *
+ *  Unfortunately Creative called the codec chip of SB as a DSP. For this
+ *  reason the /dev/dsp is reserved for digitized audio use. There is a
+ *  device for true DSP processors but it will be called something else.
+ *  In v3.0 it's /dev/sndproc but this could be a temporary solution.
+ */
+#define _dmasound_h_
+
+#include <linux/types.h>
+
+#define SND_NDEVS	256	/* Number of supported devices */
+#define SND_DEV_CTL	0	/* Control port /dev/mixer */
+#define SND_DEV_SEQ	1	/* Sequencer output /dev/sequencer (FM
+				   synthesizer and MIDI output) */
+#define SND_DEV_MIDIN	2	/* Raw midi access */
+#define SND_DEV_DSP	3	/* Digitized voice /dev/dsp */
+#define SND_DEV_AUDIO	4	/* Sparc compatible /dev/audio */
+#define SND_DEV_DSP16	5	/* Like /dev/dsp but 16 bits/sample */
+#define SND_DEV_STATUS	6	/* /dev/sndstat */
+/* #7 not in use now. Was in 2.4. Free for use after v3.0. */
+#define SND_DEV_SEQ2	8	/* /dev/sequencer, level 2 interface */
+#define SND_DEV_SNDPROC 9	/* /dev/sndproc for programmable devices */
+#define SND_DEV_PSS	SND_DEV_SNDPROC
+
+/* switch on various prinks */
+#define DEBUG_DMASOUND 1
+
+#define MAX_AUDIO_DEV	5
+#define MAX_MIXER_DEV	4
+#define MAX_SYNTH_DEV	3
+#define MAX_MIDI_DEV	6
+#define MAX_TIMER_DEV	3
+
+#define MAX_CATCH_RADIUS	10
+
+#define le2be16(x)	(((x)<<8 & 0xff00) | ((x)>>8 & 0x00ff))
+#define le2be16dbl(x)	(((x)<<8 & 0xff00ff00) | ((x)>>8 & 0x00ff00ff))
+
+#define IOCTL_IN(arg, ret) \
+	do { int error = get_user(ret, (int __user *)(arg)); \
+		if (error) return error; \
+	} while (0)
+#define IOCTL_OUT(arg, ret)	ioctl_return((int __user *)(arg), ret)
+
+static inline int ioctl_return(int __user *addr, int value)
+{
+	return value < 0 ? value : put_user(value, addr);
+}
+
+
+    /*
+     *  Configuration
+     */
+
+#undef HAS_8BIT_TABLES
+
+#if defined(CONFIG_DMASOUND_ATARI) || defined(CONFIG_DMASOUND_ATARI_MODULE) ||\
+    defined(CONFIG_DMASOUND_PAULA) || defined(CONFIG_DMASOUND_PAULA_MODULE) ||\
+    defined(CONFIG_DMASOUND_Q40) || defined(CONFIG_DMASOUND_Q40_MODULE)
+#define HAS_8BIT_TABLES
+#define MIN_BUFFERS	4
+#define MIN_BUFSIZE	(1<<12)	/* in bytes (- where does this come from ?) */
+#define MIN_FRAG_SIZE	8	/* not 100% sure about this */
+#define MAX_BUFSIZE	(1<<17)	/* Limit for Amiga is 128 kb */
+#define MAX_FRAG_SIZE	15	/* allow *4 for mono-8 => stereo-16 (for multi) */
+
+#else /* is pmac and multi is off */
+
+#define MIN_BUFFERS	2
+#define MIN_BUFSIZE	(1<<8)	/* in bytes */
+#define MIN_FRAG_SIZE	8
+#define MAX_BUFSIZE	(1<<18)	/* this is somewhat arbitrary for pmac */
+#define MAX_FRAG_SIZE	16	/* need to allow *4 for mono-8 => stereo-16 */
+#endif
+
+#define DEFAULT_N_BUFFERS 4
+#define DEFAULT_BUFF_SIZE (1<<15)
+
+    /*
+     *  Initialization
+     */
+
+extern int dmasound_init(void);
+#ifdef MODULE
+extern void dmasound_deinit(void);
+#else
+#define dmasound_deinit()	do { } while (0)
+#endif
+
+/* description of the set-up applies to either hard or soft settings */
+
+typedef struct {
+    int format;		/* AFMT_* */
+    int stereo;		/* 0 = mono, 1 = stereo */
+    int size;		/* 8/16 bit*/
+    int speed;		/* speed */
+} SETTINGS;
+
+    /*
+     *  Machine definitions
+     */
+
+typedef struct {
+    const char *name;
+    const char *name2;
+    struct module *owner;
+    void *(*dma_alloc)(unsigned int, gfp_t);
+    void (*dma_free)(void *, unsigned int);
+    int (*irqinit)(void);
+#ifdef MODULE
+    void (*irqcleanup)(void);
+#endif
+    void (*init)(void);
+    void (*silence)(void);
+    int (*setFormat)(int);
+    int (*setVolume)(int);
+    int (*setBass)(int);
+    int (*setTreble)(int);
+    int (*setGain)(int);
+    void (*play)(void);
+    void (*record)(void);		/* optional */
+    void (*mixer_init)(void);		/* optional */
+    int (*mixer_ioctl)(u_int, u_long);	/* optional */
+    int (*write_sq_setup)(void);	/* optional */
+    int (*read_sq_setup)(void);		/* optional */
+    int (*sq_open)(fmode_t);		/* optional */
+    int (*state_info)(char *, size_t);	/* optional */
+    void (*abort_read)(void);		/* optional */
+    int min_dsp_speed;
+    int max_dsp_speed;
+    int version ;
+    int hardware_afmts ;		/* OSS says we only return h'ware info */
+					/* when queried via SNDCTL_DSP_GETFMTS */
+    int capabilities ;		/* low-level reply to SNDCTL_DSP_GETCAPS */
+    SETTINGS default_hard ;	/* open() or init() should set something valid */
+    SETTINGS default_soft ;	/* you can make it look like old OSS, if you want to */
+} MACHINE;
+
+    /*
+     *  Low level stuff
+     */
+
+typedef struct {
+    ssize_t (*ct_ulaw)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t);
+    ssize_t (*ct_alaw)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t);
+    ssize_t (*ct_s8)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t);
+    ssize_t (*ct_u8)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t);
+    ssize_t (*ct_s16be)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t);
+    ssize_t (*ct_u16be)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t);
+    ssize_t (*ct_s16le)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t);
+    ssize_t (*ct_u16le)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t);
+} TRANS;
+
+struct sound_settings {
+    MACHINE mach;	/* machine dependent things */
+    SETTINGS hard;	/* hardware settings */
+    SETTINGS soft;	/* software settings */
+    SETTINGS dsp;	/* /dev/dsp default settings */
+    TRANS *trans_write;	/* supported translations */
+    int volume_left;	/* volume (range is machine dependent) */
+    int volume_right;
+    int bass;		/* tone (range is machine dependent) */
+    int treble;
+    int gain;
+    int minDev;		/* minor device number currently open */
+    spinlock_t lock;
+};
+
+extern struct sound_settings dmasound;
+
+#ifdef HAS_8BIT_TABLES
+extern char dmasound_ulaw2dma8[];
+extern char dmasound_alaw2dma8[];
+#endif
+
+    /*
+     *  Mid level stuff
+     */
+
+static inline int dmasound_set_volume(int volume)
+{
+	return dmasound.mach.setVolume(volume);
+}
+
+static inline int dmasound_set_bass(int bass)
+{
+	return dmasound.mach.setBass ? dmasound.mach.setBass(bass) : 50;
+}
+
+static inline int dmasound_set_treble(int treble)
+{
+	return dmasound.mach.setTreble ? dmasound.mach.setTreble(treble) : 50;
+}
+
+static inline int dmasound_set_gain(int gain)
+{
+	return dmasound.mach.setGain ? dmasound.mach.setGain(gain) : 100;
+}
+
+
+    /*
+     * Sound queue stuff, the heart of the driver
+     */
+
+struct sound_queue {
+    /* buffers allocated for this queue */
+    int numBufs;		/* real limits on what the user can have */
+    int bufSize;		/* in bytes */
+    char **buffers;
+
+    /* current parameters */
+    int locked ;		/* params cannot be modified when != 0 */
+    int user_frags ;		/* user requests this many */
+    int user_frag_size ;	/* of this size */
+    int max_count;		/* actual # fragments <= numBufs */
+    int block_size;		/* internal block size in bytes */
+    int max_active;		/* in-use fragments <= max_count */
+
+    /* it shouldn't be necessary to declare any of these volatile */
+    int front, rear, count;
+    int rear_size;
+    /*
+     *	The use of the playing field depends on the hardware
+     *
+     *	Atari, PMac: The number of frames that are loaded/playing
+     *
+     *	Amiga: Bit 0 is set: a frame is loaded
+     *	       Bit 1 is set: a frame is playing
+     */
+    int active;
+    wait_queue_head_t action_queue, open_queue, sync_queue;
+    int non_blocking;
+    int busy, syncing, xruns, died;
+};
+
+#define WAKE_UP(queue)		(wake_up_interruptible(&queue))
+
+extern struct sound_queue dmasound_write_sq;
+#define write_sq	dmasound_write_sq
+
+extern int dmasound_catchRadius;
+#define catchRadius	dmasound_catchRadius
+
+/* define the value to be put in the byte-swap reg in mac-io
+   when we want it to swap for us.
+*/
+#define BS_VAL 1
+
+#define SW_INPUT_VOLUME_SCALE	4
+#define SW_INPUT_VOLUME_DEFAULT	(128 / SW_INPUT_VOLUME_SCALE)
+
+extern int expand_read_bal;	/* Balance factor for reading */
+extern uint software_input_volume; /* software implemented recording volume! */
+
+#endif /* _dmasound_h_ */

sound/oss/dmasound/dmasound_paula.c

@@ -0,0 +1,739 @@
+/*
+ *  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",
+		.owner	= THIS_MODULE,
+	},
+};
+
+module_platform_driver_probe(amiga_audio_driver, amiga_audio_probe);
+
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:amiga-audio");

sound/oss/dmasound/dmasound_q40.c

@@ -0,0 +1,638 @@
+/*
+ *  linux/sound/oss/dmasound/dmasound_q40.c
+ *
+ *  Q40 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 default hard/soft settings.
+ */
+
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/soundcard.h>
+#include <linux/interrupt.h>
+
+#include <asm/uaccess.h>
+#include <asm/q40ints.h>
+#include <asm/q40_master.h>
+
+#include "dmasound.h"
+
+#define DMASOUND_Q40_REVISION 0
+#define DMASOUND_Q40_EDITION 3
+
+static int expand_bal;	/* Balance factor for expanding (not volume!) */
+static int expand_data;	/* Data for expanding */
+
+
+/*** Low level stuff *********************************************************/
+
+
+static void *Q40Alloc(unsigned int size, gfp_t flags);
+static void Q40Free(void *, unsigned int);
+static int Q40IrqInit(void);
+#ifdef MODULE
+static void Q40IrqCleanUp(void);
+#endif
+static void Q40Silence(void);
+static void Q40Init(void);
+static int Q40SetFormat(int format);
+static int Q40SetVolume(int volume);
+static void Q40PlayNextFrame(int index);
+static void Q40Play(void);
+static irqreturn_t Q40StereoInterrupt(int irq, void *dummy);
+static irqreturn_t Q40MonoInterrupt(int irq, void *dummy);
+static void Q40Interrupt(void);
+
+
+/*** Mid level stuff *********************************************************/
+
+
+
+/* userCount, frameUsed, frameLeft == byte counts */
+static ssize_t q40_ct_law(const u_char __user *userPtr, size_t userCount,
+			   u_char frame[], ssize_t *frameUsed,
+			   ssize_t frameLeft)
+{
+	char *table = dmasound.soft.format == AFMT_MU_LAW ? dmasound_ulaw2dma8: dmasound_alaw2dma8;
+	ssize_t count, used;
+	u_char *p = (u_char *) &frame[*frameUsed];
+
+	used = count = min_t(size_t, userCount, frameLeft);
+	if (copy_from_user(p,userPtr,count))
+	  return -EFAULT;
+	while (count > 0) {
+		*p = table[*p]+128;
+		p++;
+		count--;
+	}
+	*frameUsed += used ;
+	return used;
+}
+
+
+static ssize_t q40_ct_s8(const u_char __user *userPtr, size_t userCount,
+			  u_char frame[], ssize_t *frameUsed,
+			  ssize_t frameLeft)
+{
+	ssize_t count, used;
+	u_char *p = (u_char *) &frame[*frameUsed];
+
+	used = count = min_t(size_t, userCount, frameLeft);
+	if (copy_from_user(p,userPtr,count))
+	  return -EFAULT;
+	while (count > 0) {
+		*p = *p + 128;
+		p++;
+		count--;
+	}
+	*frameUsed += used;
+	return used;
+}
+
+static ssize_t q40_ct_u8(const u_char __user *userPtr, size_t userCount,
+			  u_char frame[], ssize_t *frameUsed,
+			  ssize_t frameLeft)
+{
+	ssize_t count, used;
+	u_char *p = (u_char *) &frame[*frameUsed];
+
+	used = count = min_t(size_t, userCount, frameLeft);
+	if (copy_from_user(p,userPtr,count))
+	  return -EFAULT;
+	*frameUsed += used;
+	return used;
+}
+
+
+/* a bit too complicated to optimise right now ..*/
+static ssize_t q40_ctx_law(const u_char __user *userPtr, size_t userCount,
+			    u_char frame[], ssize_t *frameUsed,
+			    ssize_t frameLeft)
+{
+	unsigned char *table = (unsigned char *)
+		(dmasound.soft.format == AFMT_MU_LAW ? dmasound_ulaw2dma8: dmasound_alaw2dma8);
+	unsigned int data = expand_data;
+	u_char *p = (u_char *) &frame[*frameUsed];
+	int bal = expand_bal;
+	int hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed;
+	int utotal, ftotal;
+
+	ftotal = frameLeft;
+	utotal = userCount;
+	while (frameLeft) {
+		u_char c;
+		if (bal < 0) {
+			if (userCount == 0)
+				break;
+			if (get_user(c, userPtr++))
+				return -EFAULT;
+			data = table[c];
+			data += 0x80;
+			userCount--;
+			bal += hSpeed;
+		}
+		*p++ = data;
+		frameLeft--;
+		bal -= sSpeed;
+	}
+	expand_bal = bal;
+	expand_data = data;
+	*frameUsed += (ftotal - frameLeft);
+	utotal -= userCount;
+	return utotal;
+}
+
+
+static ssize_t q40_ctx_s8(const u_char __user *userPtr, size_t userCount,
+			   u_char frame[], ssize_t *frameUsed,
+			   ssize_t frameLeft)
+{
+	u_char *p = (u_char *) &frame[*frameUsed];
+	unsigned int data = expand_data;
+	int bal = expand_bal;
+	int hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed;
+	int utotal, ftotal;
+
+
+	ftotal = frameLeft;
+	utotal = userCount;
+	while (frameLeft) {
+		u_char c;
+		if (bal < 0) {
+			if (userCount == 0)
+				break;
+			if (get_user(c, userPtr++))
+				return -EFAULT;
+			data = c ;
+			data += 0x80;
+			userCount--;
+			bal += hSpeed;
+		}
+		*p++ = data;
+		frameLeft--;
+		bal -= sSpeed;
+	}
+	expand_bal = bal;
+	expand_data = data;
+	*frameUsed += (ftotal - frameLeft);
+	utotal -= userCount;
+	return utotal;
+}
+
+
+static ssize_t q40_ctx_u8(const u_char __user *userPtr, size_t userCount,
+			   u_char frame[], ssize_t *frameUsed,
+			   ssize_t frameLeft)
+{
+	u_char *p = (u_char *) &frame[*frameUsed];
+	unsigned int data = expand_data;
+	int bal = expand_bal;
+	int hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed;
+	int utotal, ftotal;
+
+	ftotal = frameLeft;
+	utotal = userCount;
+	while (frameLeft) {
+		u_char c;
+		if (bal < 0) {
+			if (userCount == 0)
+				break;
+			if (get_user(c, userPtr++))
+				return -EFAULT;
+			data = c ;
+			userCount--;
+			bal += hSpeed;
+		}
+		*p++ = data;
+		frameLeft--;
+		bal -= sSpeed;
+	}
+	expand_bal = bal;
+	expand_data = data;
+	*frameUsed += (ftotal - frameLeft) ;
+	utotal -= userCount;
+	return utotal;
+}
+
+/* compressing versions */
+static ssize_t q40_ctc_law(const u_char __user *userPtr, size_t userCount,
+			    u_char frame[], ssize_t *frameUsed,
+			    ssize_t frameLeft)
+{
+	unsigned char *table = (unsigned char *)
+		(dmasound.soft.format == AFMT_MU_LAW ? dmasound_ulaw2dma8: dmasound_alaw2dma8);
+	unsigned int data = expand_data;
+	u_char *p = (u_char *) &frame[*frameUsed];
+	int bal = expand_bal;
+	int hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed;
+	int utotal, ftotal;
+ 
+	ftotal = frameLeft;
+	utotal = userCount;
+	while (frameLeft) {
+		u_char c;
+		while(bal<0) {
+			if (userCount == 0)
+				goto lout;
+			if (!(bal<(-hSpeed))) {
+				if (get_user(c, userPtr))
+					return -EFAULT;
+				data = 0x80 + table[c];
+			}
+			userPtr++;
+			userCount--;
+			bal += hSpeed;
+		}
+		*p++ = data;
+		frameLeft--;
+		bal -= sSpeed;
+	}
+ lout:
+	expand_bal = bal;
+	expand_data = data;
+	*frameUsed += (ftotal - frameLeft);
+	utotal -= userCount;
+	return utotal;
+}
+
+
+static ssize_t q40_ctc_s8(const u_char __user *userPtr, size_t userCount,
+			   u_char frame[], ssize_t *frameUsed,
+			   ssize_t frameLeft)
+{
+	u_char *p = (u_char *) &frame[*frameUsed];
+	unsigned int data = expand_data;
+	int bal = expand_bal;
+	int hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed;
+	int utotal, ftotal;
+
+	ftotal = frameLeft;
+	utotal = userCount;
+	while (frameLeft) {
+		u_char c;
+		while (bal < 0) {
+			if (userCount == 0)
+				goto lout;
+			if (!(bal<(-hSpeed))) {
+				if (get_user(c, userPtr))
+					return -EFAULT;
+				data = c + 0x80;
+			}
+			userPtr++;
+			userCount--;
+			bal += hSpeed;
+		}
+		*p++ = data;
+		frameLeft--;
+		bal -= sSpeed;
+	}
+ lout:
+	expand_bal = bal;
+	expand_data = data;
+	*frameUsed += (ftotal - frameLeft);
+	utotal -= userCount;
+	return utotal;
+}
+
+
+static ssize_t q40_ctc_u8(const u_char __user *userPtr, size_t userCount,
+			   u_char frame[], ssize_t *frameUsed,
+			   ssize_t frameLeft)
+{
+	u_char *p = (u_char *) &frame[*frameUsed];
+	unsigned int data = expand_data;
+	int bal = expand_bal;
+	int hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed;
+	int utotal, ftotal;
+
+	ftotal = frameLeft;
+	utotal = userCount;
+	while (frameLeft) {
+		u_char c;
+		while (bal < 0) {
+			if (userCount == 0)
+				goto lout;
+			if (!(bal<(-hSpeed))) {
+				if (get_user(c, userPtr))
+					return -EFAULT;
+				data = c ;
+			}
+			userPtr++;
+			userCount--;
+			bal += hSpeed;
+		}
+		*p++ = data;
+		frameLeft--;
+		bal -= sSpeed;
+	}
+ lout:
+	expand_bal = bal;
+	expand_data = data;
+	*frameUsed += (ftotal - frameLeft) ;
+	utotal -= userCount;
+	return utotal;
+}
+
+
+static TRANS transQ40Normal = {
+	q40_ct_law, q40_ct_law, q40_ct_s8, q40_ct_u8, NULL, NULL, NULL, NULL
+};
+
+static TRANS transQ40Expanding = {
+	q40_ctx_law, q40_ctx_law, q40_ctx_s8, q40_ctx_u8, NULL, NULL, NULL, NULL
+};
+
+static TRANS transQ40Compressing = {
+	q40_ctc_law, q40_ctc_law, q40_ctc_s8, q40_ctc_u8, NULL, NULL, NULL, NULL
+};
+
+
+/*** Low level stuff *********************************************************/
+
+static void *Q40Alloc(unsigned int size, gfp_t flags)
+{
+         return kmalloc(size, flags); /* change to vmalloc */
+}
+
+static void Q40Free(void *ptr, unsigned int size)
+{
+	kfree(ptr);
+}
+
+static int __init Q40IrqInit(void)
+{
+	/* Register interrupt handler. */
+	if (request_irq(Q40_IRQ_SAMPLE, Q40StereoInterrupt, 0,
+		    "DMA sound", Q40Interrupt))
+		return 0;
+
+	return(1);
+}
+
+
+#ifdef MODULE
+static void Q40IrqCleanUp(void)
+{
+        master_outb(0,SAMPLE_ENABLE_REG);
+	free_irq(Q40_IRQ_SAMPLE, Q40Interrupt);
+}
+#endif /* MODULE */
+
+
+static void Q40Silence(void)
+{
+        master_outb(0,SAMPLE_ENABLE_REG);
+	*DAC_LEFT=*DAC_RIGHT=127;
+}
+
+static char *q40_pp;
+static unsigned int q40_sc;
+
+static void Q40PlayNextFrame(int index)
+{
+	u_char *start;
+	u_long size;
+	u_char speed;
+	int error;
+
+	/* used by Q40Play() 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);
+
+	q40_pp=start;
+	q40_sc=size;
+
+	write_sq.front = (write_sq.front+1) % write_sq.max_count;
+	write_sq.active++;
+
+	speed=(dmasound.hard.speed==10000 ? 0 : 1);
+
+	master_outb( 0,SAMPLE_ENABLE_REG);
+	free_irq(Q40_IRQ_SAMPLE, Q40Interrupt);
+	if (dmasound.soft.stereo)
+		error = request_irq(Q40_IRQ_SAMPLE, Q40StereoInterrupt, 0,
+				    "Q40 sound", Q40Interrupt);
+	  else
+		error = request_irq(Q40_IRQ_SAMPLE, Q40MonoInterrupt, 0,
+				    "Q40 sound", Q40Interrupt);
+	if (error && printk_ratelimit())
+		pr_err("Couldn't register sound interrupt\n");
+
+	master_outb( speed, SAMPLE_RATE_REG);
+	master_outb( 1,SAMPLE_CLEAR_REG);
+	master_outb( 1,SAMPLE_ENABLE_REG);
+}
+
+static void Q40Play(void)
+{
+        unsigned long flags;
+
+	if (write_sq.active || write_sq.count<=0 ) {
+		/* There's already a frame loaded */
+		return;
+	}
+
+	/* nothing in the queue */
+	if (write_sq.count <= 1 && 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?
+		  */
+	         return;
+	}
+	spin_lock_irqsave(&dmasound.lock, flags);
+	Q40PlayNextFrame(1);
+	spin_unlock_irqrestore(&dmasound.lock, flags);
+}
+
+static irqreturn_t Q40StereoInterrupt(int irq, void *dummy)
+{
+	spin_lock(&dmasound.lock);
+        if (q40_sc>1){
+            *DAC_LEFT=*q40_pp++;
+	    *DAC_RIGHT=*q40_pp++;
+	    q40_sc -=2;
+	    master_outb(1,SAMPLE_CLEAR_REG);
+	}else Q40Interrupt();
+	spin_unlock(&dmasound.lock);
+	return IRQ_HANDLED;
+}
+static irqreturn_t Q40MonoInterrupt(int irq, void *dummy)
+{
+	spin_lock(&dmasound.lock);
+        if (q40_sc>0){
+            *DAC_LEFT=*q40_pp;
+	    *DAC_RIGHT=*q40_pp++;
+	    q40_sc --;
+	    master_outb(1,SAMPLE_CLEAR_REG);
+	}else Q40Interrupt();
+	spin_unlock(&dmasound.lock);
+	return IRQ_HANDLED;
+}
+static void Q40Interrupt(void)
+{
+	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);
+		   master_outb(0,SAMPLE_ENABLE_REG); /* better safe */
+		   goto exit;
+	} else write_sq.active=0;
+	write_sq.count--;
+	Q40Play();
+
+	if (q40_sc<2)
+	      { /* there was nothing to play, disable irq */
+		master_outb(0,SAMPLE_ENABLE_REG);
+		*DAC_LEFT=*DAC_RIGHT=127;
+	      }
+	WAKE_UP(write_sq.action_queue);
+
+ exit:
+	master_outb(1,SAMPLE_CLEAR_REG);
+}
+
+
+static void Q40Init(void)
+{
+	int i, idx;
+	const int freq[] = {10000, 20000};
+
+	/* search a frequency that fits into the allowed error range */
+
+	idx = -1;
+	for (i = 0; i < 2; i++)
+		if ((100 * abs(dmasound.soft.speed - freq[i]) / freq[i]) <= catchRadius)
+			idx = i;
+
+	dmasound.hard = dmasound.soft;
+	/*sound.hard.stereo=1;*/ /* no longer true */
+	dmasound.hard.size=8;
+
+	if (idx > -1) {
+		dmasound.soft.speed = freq[idx];
+		dmasound.trans_write = &transQ40Normal;
+	} else
+		dmasound.trans_write = &transQ40Expanding;
+
+	Q40Silence();
+
+	if (dmasound.hard.speed > 20200) {
+		/* squeeze the sound, we do that */
+		dmasound.hard.speed = 20000;
+		dmasound.trans_write = &transQ40Compressing;
+	} else if (dmasound.hard.speed > 10000) {
+		dmasound.hard.speed = 20000;
+	} else {
+		dmasound.hard.speed = 10000;
+	}
+	expand_bal = -dmasound.soft.speed;
+}
+
+
+static int Q40SetFormat(int format)
+{
+	/* Q40 sound supports only 8bit modes */
+
+	switch (format) {
+	case AFMT_QUERY:
+		return(dmasound.soft.format);
+	case AFMT_MU_LAW:
+	case AFMT_A_LAW:
+	case AFMT_S8:
+	case AFMT_U8:
+		break;
+	default:
+		format = AFMT_S8;
+	}
+
+	dmasound.soft.format = format;
+	dmasound.soft.size = 8;
+	if (dmasound.minDev == SND_DEV_DSP) {
+		dmasound.dsp.format = format;
+		dmasound.dsp.size = 8;
+	}
+	Q40Init();
+
+	return(format);
+}
+
+static int Q40SetVolume(int volume)
+{
+    return 0;
+}
+
+
+/*** Machine definitions *****************************************************/
+
+static SETTINGS def_hard = {
+	.format	= AFMT_U8,
+	.stereo	= 0,
+	.size	= 8,
+	.speed	= 10000
+} ;
+
+static SETTINGS def_soft = {
+	.format	= AFMT_U8,
+	.stereo	= 0,
+	.size	= 8,
+	.speed	= 8000
+} ;
+
+static MACHINE machQ40 = {
+	.name		= "Q40",
+	.name2		= "Q40",
+	.owner		= THIS_MODULE,
+	.dma_alloc	= Q40Alloc,
+	.dma_free	= Q40Free,
+	.irqinit	= Q40IrqInit,
+#ifdef MODULE
+	.irqcleanup	= Q40IrqCleanUp,
+#endif /* MODULE */
+	.init		= Q40Init,
+	.silence	= Q40Silence,
+	.setFormat	= Q40SetFormat,
+	.setVolume	= Q40SetVolume,
+	.play		= Q40Play,
+ 	.min_dsp_speed	= 10000,
+	.version	= ((DMASOUND_Q40_REVISION<<8) | DMASOUND_Q40_EDITION),
+	.hardware_afmts	= AFMT_U8, /* h'ware-supported formats *only* here */
+	.capabilities	= DSP_CAP_BATCH  /* As per SNDCTL_DSP_GETCAPS */
+};
+
+
+/*** Config & Setup **********************************************************/
+
+
+static int __init dmasound_q40_init(void)
+{
+	if (MACH_IS_Q40) {
+	    dmasound.mach = machQ40;
+	    dmasound.mach.default_hard = def_hard ;
+	    dmasound.mach.default_soft = def_soft ;
+	    return dmasound_init();
+	} else
+	    return -ENODEV;
+}
+
+static void __exit dmasound_q40_cleanup(void)
+{
+	dmasound_deinit();
+}
+
+module_init(dmasound_q40_init);
+module_exit(dmasound_q40_cleanup);
+
+MODULE_DESCRIPTION("Q40/Q60 sound driver");
+MODULE_LICENSE("GPL");