// 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 = __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");