/**

 * $Id: audio.c,v 1.1 2006-01-10 13:56:54 nkeynes Exp $

 * 

 * Audio mixer core. Combines all the active streams into a single sound

 * buffer for output. 

 *

 * 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.

 */

#include "aica/aica.h"

#include "aica/audio.h"

#include "glib/gmem.h"

#include "dream.h"

#include <assert.h>

#include <string.h>

#define NUM_BUFFERS 3

#define MS_PER_BUFFER 4000

#define BUFFER_EMPTY   0

#define BUFFER_WRITING 1

#define BUFFER_FULL    2

struct audio_state {

    audio_buffer_t output_buffers[NUM_BUFFERS];

    int write_buffer;

    int read_buffer;

    uint32_t output_format;

    uint32_t output_rate;

    uint32_t output_sample_size;

    struct audio_channel channels[64];

} audio;

audio_driver_t audio_driver = NULL;

#define NEXT_BUFFER() ((audio.write_buffer == NUM_BUFFERS-1) ? 0 : audio.write_buffer+1)

extern char *arm_mem;

/**

 * Set the output driver, sample rate and format. Also initializes the 

 * output buffers, flushing any current data and reallocating as 

 * necessary.

 */

void audio_set_output( audio_driver_t driver, 

		       uint32_t samplerate, int format )

{

    uint32_t bytes_per_sample = 1;

    uint32_t samples_per_buffer;

    int i;

    if( format & AUDIO_FMT_16BIT )

	bytes_per_sample = 2;

    if( format & AUDIO_FMT_STEREO )

	bytes_per_sample <<= 1;

    if( samplerate == audio.output_rate &&

	bytes_per_sample == audio.output_sample_size )

	return;

    samples_per_buffer = (samplerate * MS_PER_BUFFER / 1000);

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

	if( audio.output_buffers[i] != NULL )

	    free(audio.output_buffers[i]);

	audio.output_buffers[i] = g_malloc0( sizeof(struct audio_buffer) + samples_per_buffer * bytes_per_sample );

	audio.output_buffers[i]->length = samples_per_buffer;

	audio.output_buffers[i]->posn = 0;

	audio.output_buffers[i]->status = BUFFER_EMPTY;

    }

    audio.output_format = format;

    audio.output_rate = samplerate;

    audio.output_sample_size = bytes_per_sample;

    audio.write_buffer = 0;

    audio.read_buffer = 0;

    if( driver == NULL )

	driver = &null_audio_driver;

    audio_driver = driver;

    audio_driver->set_output_format( samplerate, format );

}

/**

 * Mark the current write buffer as full and prepare the next buffer for

 * writing. Returns the next buffer to write to.

 * If all buffers are full, returns NULL.

 */

audio_buffer_t audio_next_write_buffer( )

{

    audio_buffer_t result = NULL;

    audio_buffer_t current = audio.output_buffers[audio.write_buffer];

    current->status = BUFFER_FULL;

    if( audio.read_buffer == audio.write_buffer &&

	audio_driver->process_buffer( current ) ) {

	audio_next_read_buffer();

    }

    audio.write_buffer = NEXT_BUFFER();

    result = audio.output_buffers[audio.write_buffer];

    if( result->status == BUFFER_FULL )

	return NULL;

    else {

	result->status = BUFFER_WRITING;

	return result;

    }

}

/**

 * Mark the current read buffer as empty and return the next buffer for

 * reading. If there is no next buffer yet, returns NULL.

 */

audio_buffer_t audio_next_read_buffer( )

{

    audio_buffer_t current = audio.output_buffers[audio.read_buffer];

    assert( current->status == BUFFER_FULL );

    current->status = BUFFER_EMPTY;

    current->posn = 0;

    audio.read_buffer++;

    if( audio.read_buffer == NUM_BUFFERS )

	audio.read_buffer = 0;

    current = audio.output_buffers[audio.read_buffer];

    if( current->status == BUFFER_FULL )

	return current;

    else return NULL;

}

/*************************** ADPCM ***********************************/

/**

 * The following section borrows heavily from ffmpeg, which is

 * copyright (c) 2001-2003 by the fine folks at the ffmpeg project,

 * distributed under the GPL version 2 or later.

 */

#define CLAMP_TO_SHORT(value) \

if (value > 32767) \

    value = 32767; \

else if (value < -32768) \

    value = -32768; \

static const int yamaha_indexscale[] = {

    230, 230, 230, 230, 307, 409, 512, 614,

    230, 230, 230, 230, 307, 409, 512, 614

};

static const int yamaha_difflookup[] = {

    1, 3, 5, 7, 9, 11, 13, 15,

    -1, -3, -5, -7, -9, -11, -13, -15

};

static inline short adpcm_yamaha_decode_nibble( audio_channel_t c, 

						unsigned char nibble )

{

    if( c->adpcm_step == 0 ) {

        c->adpcm_predict = 0;

        c->adpcm_step = 127;

    }

    c->adpcm_predict += (c->adpcm_step * yamaha_difflookup[nibble]) >> 3;

    CLAMP_TO_SHORT(c->adpcm_predict);

    c->adpcm_step = (c->adpcm_step * yamaha_indexscale[nibble]) >> 8;

    c->adpcm_step = CLAMP(c->adpcm_step, 127, 24567);

    return c->adpcm_predict;

}

/*************************** Sample mixer *****************************/

/**

 * Mix a single output sample.

 */

void audio_mix_sample( )

{

    int i, j;

    int32_t result_left = 0, result_right = 0;

    for( i=0; i < 64; i++ ) {

	audio_channel_t channel = &audio.channels[i];

	if( channel->active ) {

	    int32_t sample;

	    switch( channel->sample_format ) {

	    case AUDIO_FMT_16BIT:

		sample = *(int16_t *)(arm_mem + channel->posn + channel->start);

		break;

	    case AUDIO_FMT_8BIT:

		sample = (*(int8_t *)(arm_mem + channel->posn + channel->start)) << 8;

		break;

	    case AUDIO_FMT_16BIT|AUDIO_FMT_UNSIGNED:

		sample = (int8_t)((*(uint16_t *)(arm_mem + channel->posn + channel->start)) - 0x8000);

		break;

	    case AUDIO_FMT_8BIT|AUDIO_FMT_UNSIGNED:

		sample = (int8_t)((*(uint8_t *)(arm_mem + channel->posn + channel->start)) - 0x80);

		break;

	    case AUDIO_FMT_ADPCM:

		sample = (int16_t)channel->adpcm_predict;

	    default:

		sample = 0; /* Unsupported */

	    }

	    result_left += sample * channel->vol_left;

	    result_right += sample * channel->vol_right;

	    channel->posn_left += channel->sample_rate;

	    while( channel->posn_left > audio.output_rate ) {

		channel->posn_left -= audio.output_rate;

		if( channel->sample_format == AUDIO_FMT_ADPCM &&

		    channel->adpcm_nibble == 0 ) {

		    uint8_t data = *(uint8_t *)(arm_mem + channel->posn + channel->start);

		    adpcm_yamaha_decode_nibble( channel, (data >> 4) & 0x0F );

		    channel->adpcm_nibble = 1;

		    continue;

		}

		channel->posn++;

		if( channel->loop_count != 0 && 

		    channel->posn >= channel->loop_end ) {

		    channel->posn = channel->loop_start;

		    if( channel->loop_count != -1 )

			channel->loop_count --;

		} else if( channel->posn >= channel->end ) {

		    audio_stop_channel( i );

		    break;

		}

		if( channel->sample_format == AUDIO_FMT_ADPCM ) {

		    uint8_t data = *(uint8_t *)(arm_mem + channel->posn + channel->start);

		    adpcm_yamaha_decode_nibble( channel, data & 0x0F );

		    channel->adpcm_nibble = 0;

		}

	    }

	}

    }

    /* Down-render to the final output format */

    audio_buffer_t buf = 

	audio.output_buffers[audio.write_buffer];

    if( audio.output_format & AUDIO_FMT_16BIT ) {

	uint16_t *data = (uint16_t *)&buf->data[buf->posn*audio.output_sample_size];

	*data++ = (int16_t)(result_left >> 8);

	*data++ = (int16_t)(result_right >> 8);

    } else {

	audio_buffer_t buf = 

	    audio.output_buffers[audio.write_buffer];

	uint8_t *data = (uint8_t *)&buf->data[buf->posn*audio.output_sample_size];

	*data++ = (int8_t)(result_left >> 22);

	*data++ = (int8_t)(result_right >>22);

    }

    buf->posn++;

    if( buf->posn == buf->length ) {

	audio_next_write_buffer();

    }

}

/********************** Internal AICA calls ***************************/

audio_channel_t audio_get_channel( int channel ) 

{

    return &audio.channels[channel];

}

void audio_stop_channel( int channel ) 

{

    audio.channels[channel].active = FALSE;

}

void audio_start_channel( int channel )

{

    audio.channels[channel].posn = 0;

    audio.channels[channel].posn_left = 0;

    audio.channels[channel].adpcm_nibble = 0;

    audio.channels[channel].adpcm_step = 0;

    audio.channels[channel].adpcm_predict = 0;

    audio.channels[channel].active = TRUE;

    DEBUG("Channel %d on", channel );

}