// SPDX-License-Identifier: GPL-2.0-only
/*
* 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 <linux/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 void __exit amiga_audio_remove(struct platform_device *pdev)
{
dmasound_deinit();
}
/*
* amiga_audio_remove() lives in .exit.text. For drivers registered via
* module_platform_driver_probe() this is ok because they cannot get unbound at
* runtime. So mark the driver struct with __refdata to prevent modpost
* triggering a section mismatch warning.
*/
static struct platform_driver amiga_audio_driver __refdata = {
.remove_new = __exit_p(amiga_audio_remove),
.driver = {
.name = "amiga-audio",
},
};
module_platform_driver_probe(amiga_audio_driver, amiga_audio_probe);
MODULE_DESCRIPTION("Amiga Paula DMA Sound Driver");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:amiga-audio");
Codebase Browser
linux