dmasound_paula.c

linux-2.6.15.6

/*
 *  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/config.h>
#include <linux/mm.h>
#include <linux/init.h>
#include <linux/ioport.h>
#include <linux/soundcard.h>
#include <linux/interrupt.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

   /*
    *	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, struct pt_regs *fp);

#ifdef CONFIG_HEARTBEAT

    /*
     *  Heartbeat interferes with sound since the 7 kHz low-pass filter and the
     *  power LED are controlled by the same line.
     */

#ifdef CONFIG_APUS
#define mach_heartbeat	ppc_md.heartbeat
#endif

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 *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 *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 *userPtr, size_t userCount,	\
			u_char frame[], ssize_t *frameUsed,		\
			ssize_t frameLeft)				\
{									\
	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, ((u_short *)userPtr)++))	\
				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, ((u_short *)userPtr)++))	\
				return -EFAULT;				\
			data = convsample(data);			\
			*lefth++ = data>>8;				\
			*leftl++ = (data>>2) & 0x3f;			\
			if (get_user(data, ((u_short *)userPtr)++))	\
				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) {