/**

 * $Id: audio.c,v 1.8 2007-10-09 08:11:51 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 50

#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.

 */

gboolean audio_set_driver( audio_driver_t driver, 

			   uint32_t samplerate, int format )

{

    uint32_t bytes_per_sample = 1;

    uint32_t samples_per_buffer;

    int i;

    if( audio_driver == NULL || driver != NULL ) {

	if( driver == NULL  )

	    driver = &audio_null_driver;

	if( driver != audio_driver ) {	

	    if( !driver->set_output_format( samplerate, format ) )

		return FALSE;

	    audio_driver = driver;

	}

    }

    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 TRUE;

    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 * bytes_per_sample;

	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;

    return TRUE;

}

/**

 * 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_samples( int num_samples )

{

    int i, j;

    int32_t result_buf[num_samples][2];

    memset( &result_buf, 0, sizeof(result_buf) );

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

	audio_channel_t channel = &audio.channels[i];

	if( channel->active ) {

	    int32_t sample;

	    int vol_left = (channel->vol * (32 - channel->pan)) >> 5;

	    int vol_right = (channel->vol * (channel->pan + 1)) >> 5;

	    switch( channel->sample_format ) {

	    case AUDIO_FMT_16BIT:

		for( j=0; j<num_samples; j++ ) {

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

		    result_buf[j][0] += sample * vol_left;

		    result_buf[j][1] += sample * vol_right;

		    channel->posn_left += channel->sample_rate;

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

			channel->posn_left -= audio.output_rate;

			channel->posn++;

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

			    if( channel->loop )

				channel->posn = channel->loop_start;

			    else {

				audio_stop_channel(i);

				j = num_samples;

				break;

			    }

			}

		    }

		}

		break;

	    case AUDIO_FMT_8BIT:

		for( j=0; j<num_samples; j++ ) {

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

		    result_buf[j][0] += sample * vol_left;

		    result_buf[j][1] += sample * vol_right;

		    channel->posn_left += channel->sample_rate;

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

			channel->posn_left -= audio.output_rate;

			channel->posn++;

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

			    if( channel->loop )

				channel->posn = channel->loop_start;

			    else {

				audio_stop_channel(i);

				j = num_samples;

				break;

			    }

			}

		    }

		}

		break;

	    case AUDIO_FMT_ADPCM:

		for( j=0; j<num_samples; j++ ) {

		    sample = (int16_t)channel->adpcm_predict;

		    result_buf[j][0] += sample * vol_left;

		    result_buf[j][1] += sample * vol_right;

		    channel->posn_left += channel->sample_rate;

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

			channel->posn_left -= audio.output_rate;

			if( 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;

			} else {

			    channel->posn++;

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

				if( channel->loop )

				    channel->posn = channel->loop_start;

				else

				    audio_stop_channel(i);

				break;

			    }

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

			    adpcm_yamaha_decode_nibble( channel, data & 0x0F );

			    channel->adpcm_nibble = 0;

			}

		    }

		}

		break;

	    default:

		break;

	    }

	}

    }

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

    if( audio.output_format & AUDIO_FMT_16BIT ) {

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

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

	for( j=0; j < num_samples; j++ ) {

	    *data++ = (int16_t)(result_buf[j][0] >> 6);

	    *data++ = (int16_t)(result_buf[j][1] >> 6);	

	    buf->posn += 4;

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

		audio_next_write_buffer();

		buf = audio.output_buffers[audio.write_buffer];

		data = (uint16_t *)&buf->data[0];

	    }

	}

    } else {

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

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

	for( j=0; j < num_samples; j++ ) {

	    *data++ = (uint8_t)(result_buf[j][0] >> 16);

	    *data++ = (uint8_t)(result_buf[j][1] >> 16);	

	    buf->posn += 2;

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

		audio_next_write_buffer();

		buf = audio.output_buffers[audio.write_buffer];

		data = (uint8_t *)&buf->data[0];

	    }

	}

    }

}

/********************** 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;

}