/**

 * $Id: aica.c,v 1.12 2006-01-12 11:30:19 nkeynes Exp $

 * 

 * This is the core sound system (ie the bit which does the actual work)

 *

 * Copyright (c) 2005 Nathan Keynes.

 *

 * This program is free software; you can redistribute it and/or modify

 * it under the terms of the GNU General Public License as published by

 * the Free Software Foundation; either version 2 of the License, or

 * (at your option) any later version.

 *

 * This program is distributed in the hope that it will be useful,

 * but WITHOUT ANY WARRANTY; without even the implied warranty of

 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the

 * GNU General Public License for more details.

 */

#define MODULE aica_module

#include "dream.h"

#include "dreamcast.h"

#include "mem.h"

#include "aica.h"

#include "armcore.h"

#include "audio.h"

#define MMIO_IMPL

#include "aica.h"

MMIO_REGION_READ_DEFFN( AICA0 )

MMIO_REGION_READ_DEFFN( AICA1 )

MMIO_REGION_READ_DEFFN( AICA2 )

void aica_init( void );

void aica_reset( void );

void aica_start( void );

void aica_stop( void );

void aica_save_state( FILE *f );

int aica_load_state( FILE *f );

uint32_t aica_run_slice( uint32_t );

struct dreamcast_module aica_module = { "AICA", aica_init, aica_reset, 

					aica_start, aica_run_slice, aica_stop,

					aica_save_state, aica_load_state };

/**

 * Initialize the AICA subsystem. Note requires that 

 */

void aica_init( void )

{

    register_io_regions( mmio_list_spu );

    MMIO_NOTRACE(AICA0);

    MMIO_NOTRACE(AICA1);

    arm_mem_init();

    aica_reset();

    audio_set_output( &esd_audio_driver, 44100, AUDIO_FMT_16BIT|AUDIO_FMT_STEREO );

}

void aica_reset( void )

{

    arm_reset();

    aica_event(2); /* Pre-deliver a timer interrupt */

}

void aica_start( void )

{

}

/**

 * Keep track of what we've done so far this second, to try to keep the

 * precision of samples/second.

 */

int samples_done = 0;

uint32_t nanosecs_done = 0;

uint32_t aica_run_slice( uint32_t nanosecs )

{

    /* Run arm instructions */

    int reset = MMIO_READ( AICA2, AICA_RESET );

    if( (reset & 1) == 0 ) { /* Running */

	int num_samples = (nanosecs_done + nanosecs) / AICA_SAMPLE_RATE - samples_done;

	num_samples = arm_run_slice( num_samples );

	audio_mix_samples( num_samples );

	samples_done += num_samples;

	nanosecs_done += nanosecs;

    }

    if( nanosecs_done > 1000000000 ) {

	samples_done -= AICA_SAMPLE_RATE;

	nanosecs_done -= 1000000000;

    }

    return nanosecs;

}

void aica_stop( void )

{

}

void aica_save_state( FILE *f )

{

    arm_save_state( f );

}

int aica_load_state( FILE *f )

{

    return arm_load_state( f );

}

int aica_event_pending = 0;

int aica_clear_count = 0;

/* Note: This is probably not necessarily technically correct but it should

 * work in the meantime.

 */

void aica_event( int event )

{

    if( aica_event_pending == 0 )

	armr.int_pending |= CPSR_F;

    aica_event_pending |= (1<<event);

    int pending = MMIO_READ( AICA2, AICA_IRQ );

    if( pending == 0 || event < pending )

	MMIO_WRITE( AICA2, AICA_IRQ, event );

}

void aica_clear_event( )

{

    aica_clear_count++;

    if( aica_clear_count == 4 ) {

	int i;

	aica_clear_count = 0;

	for( i=0; i<8; i++ ) {

	    if( aica_event_pending & (1<<i) ) {

		aica_event_pending &= ~(1<<i);

		break;

	    }

	}

	for( ;i<8; i++ ) {

	    if( aica_event_pending & (1<<i) ) {

		MMIO_WRITE( AICA2, AICA_IRQ, i );

		break;

	    }

	}

	if( aica_event_pending == 0 )

	    armr.int_pending &= ~CPSR_F;

    }

}

/** Channel register structure:

 * 00  4  Channel config

 * 04  4  Waveform address lo (16 bits)

 * 08  4  Loop start address

 * 0C  4  Loop end address

 * 10  4  Volume envelope

 * 14  4  Init to 0x1F

 * 18  4  Frequency (floating point)

 * 1C  4  ?? 

 * 20  4  ??

 * 24  1  Pan

 * 25  1  ??

 * 26  

 * 27  

 * 28  1  ??

 * 29  1  Volume

 * 2C

 * 30

 * 

/* Write to channels 0-31 */

void mmio_region_AICA0_write( uint32_t reg, uint32_t val )

{

    MMIO_WRITE( AICA0, reg, val );

    aica_write_channel( reg >> 7, reg % 128, val );

    //    DEBUG( "AICA0 Write %08X => %08X", val, reg );

}

/* Write to channels 32-64 */

void mmio_region_AICA1_write( uint32_t reg, uint32_t val )

{

    MMIO_WRITE( AICA1, reg, val );

    aica_write_channel( (reg >> 7) + 32, reg % 128, val );

    // DEBUG( "AICA1 Write %08X => %08X", val, reg );

}

/**

 * AICA control registers 

 */

void mmio_region_AICA2_write( uint32_t reg, uint32_t val )

{

    uint32_t tmp;

    switch( reg ) {

    case AICA_RESET:

	tmp = MMIO_READ( AICA2, AICA_RESET );

	if( (tmp & 1) == 1 && (val & 1) == 0 ) {

	    /* ARM enabled - execute a core reset */

	    DEBUG( "ARM enabled" );

	    arm_reset();

	    samples_done = 0;

	    nanosecs_done = 0;

	} else if( (tmp&1) == 0 && (val&1) == 1 ) {

	    DEBUG( "ARM disabled" );

	}

	MMIO_WRITE( AICA2, AICA_RESET, val );

	break;

    case AICA_IRQCLEAR:

	aica_clear_event();

	break;

    default:

	MMIO_WRITE( AICA2, reg, val );

	break;

    }

}

/**

 * Translate the channel frequency to a sample rate. The frequency is a

 * 14-bit floating point number, where bits 0..9 is the mantissa,

 * 11..14 is the signed exponent (-8 to +7). Bit 10 appears to

 * be unused.

 *

 * @return sample rate in samples per second.

 */

uint32_t aica_frequency_to_sample_rate( uint32_t freq )

{

    uint32_t exponent = (freq & 0x3800) >> 11;

    uint32_t mantissa = freq & 0x03FF;

    if( freq & 0x4000 ) {

	/* neg exponent - rate < 44100 */

	exponent = 8 - exponent;

	return (44100 >> exponent) +

	    ((44100 * mantissa) >> (10+exponent));

    } else {

	/* pos exponent - rate > 44100 */

	return (44100 << exponent) +

	    ((44100 * mantissa) >> (10-exponent));

    }

}

void aica_write_channel( int channelNo, uint32_t reg, uint32_t val ) 

{

    val &= 0x0000FFFF;

    audio_channel_t channel = audio_get_channel(channelNo);

    switch( reg ) {

    case 0x00: /* Config + high address bits*/

	channel->start = (channel->start & 0xFFFF) | ((val&0x1F) << 16);

	if( val & 0x200 ) 

	    channel->loop = TRUE;

	else 

	    channel->loop = FALSE;

	switch( (val >> 7) & 0x03 ) {

	case 0:

	    channel->sample_format = AUDIO_FMT_16BIT;

	    break;

	case 1:

	    channel->sample_format = AUDIO_FMT_8BIT;

	    break;

	case 2:

	case 3:

	    channel->sample_format = AUDIO_FMT_ADPCM;

	    break;

	}

	switch( (val >> 14) & 0x03 ) {

	case 2: 

	    audio_stop_channel( channelNo ); 

	    break;

	case 3: 

	    audio_start_channel( channelNo ); 

	    break;

	default:

	    break;

	    /* Hrmm... */

	}

	break;

    case 0x04: /* Low 16 address bits */

	channel->start = (channel->start & 0x001F0000) | val;

	break;

    case 0x08: /* Loop start */

	channel->loop_start = val;

	break;

    case 0x0C: /* End */

	channel->end = val;

	break;

    case 0x10: /* Envelope register 1 */

	break;

    case 0x14: /* Envelope register 2 */

	break;

    case 0x18: /* Frequency */

	channel->sample_rate = aica_frequency_to_sample_rate ( val );

	break;

    case 0x1C: /* ??? */

    case 0x20: /* ??? */

    case 0x24: /* Volume? /pan */

	break;

    case 0x28: /* Volume */

	channel->vol_left = channel->vol_right = val & 0xFF;

	break;

    default: /* ??? */

	break;

    }

}