/**

  * $Id: audio.c,v 1.3 2006-01-16 11:23:05 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 100

 #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 * 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;

     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_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;

 	    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 * channel->vol_left;

 		    result_buf[j][1] += sample * channel->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 * channel->vol_left;

 		    result_buf[j][1] += sample * channel->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 * channel->vol_left;

 		    result_buf[j][1] += 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->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] >> 8);

 	    *data++ = (int16_t)(result_buf[j][1] >> 8);	

 	    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;

 }