/**

 * $Id$

 * 

 * 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>

extern struct audio_driver audio_null_driver;

extern struct audio_driver audio_osx_driver;

extern struct audio_driver audio_pulse_driver;

extern struct audio_driver audio_esd_driver;

extern struct audio_driver audio_alsa_driver;

audio_driver_t audio_driver_list[] = {

#ifdef HAVE_CORE_AUDIO

        &audio_osx_driver,

#endif

#ifdef HAVE_PULSE

        &audio_pulse_driver,

#endif

#ifdef HAVE_ESOUND

        &audio_esd_driver,

#endif

#ifdef HAVE_ALSA

        &audio_alsa_driver,

#endif

        &audio_null_driver,

        NULL };

#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[AUDIO_CHANNEL_COUNT];

} 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;

/**

 * Preserve audio channel state only - don't bother saving the buffers

 */

void audio_save_state( FILE *f )

{

    fwrite( &audio.channels[0], sizeof(struct audio_channel), AUDIO_CHANNEL_COUNT, f );

}

int audio_load_state( FILE *f )

{

    int read = fread( &audio.channels[0], sizeof(struct audio_channel), AUDIO_CHANNEL_COUNT, f );

    return (read == AUDIO_CHANNEL_COUNT ? 0 : -1 );

}

audio_driver_t get_audio_driver_by_name( const char *name )

{

    int i;

    if( name == NULL ) {

        return audio_driver_list[0];

    }

    for( i=0; audio_driver_list[i] != NULL; i++ ) {

        if( strcasecmp( audio_driver_list[i]->name, name ) == 0 ) {

            return audio_driver_list[i];

        }

    }

    return NULL;

}

void print_audio_drivers( FILE * out )

{

    int i;

    fprintf( out, "Available audio drivers:\n" );

    for( i=0; audio_driver_list[i] != NULL; i++ ) {

        fprintf( out, "  %-8s %s\n", audio_driver_list[i]->name,

                gettext(audio_driver_list[i]->description) );

    }

}

audio_driver_t audio_init_driver( const char *preferred_driver )

{

    audio_driver_t audio_driver = get_audio_driver_by_name(preferred_driver);

    if( audio_driver == NULL ) {

        ERROR( "Audio driver '%s' not found, aborting.", preferred_driver );

        exit(2);

    } else if( audio_set_driver( audio_driver ) == FALSE ) {

        ERROR( "Failed to initialize audio driver '%s', using null driver", 

                audio_driver->name );

        audio_set_driver( &audio_null_driver );

    }    

}

/**

 * 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 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->init() )

                return FALSE;

            audio_driver = driver;

        }

    }

    switch( driver->sample_format & AUDIO_FMT_SAMPLE_MASK ) {

    case AUDIO_FMT_8BIT:

        bytes_per_sample = 1;

        break;

    case AUDIO_FMT_16BIT:

        bytes_per_sample = 2;

        break;

    case AUDIO_FMT_FLOAT:

        bytes_per_sample = 4;

        break;

    }

    if( driver->sample_format & AUDIO_FMT_STEREO )

        bytes_per_sample <<= 1;

    if( driver->sample_rate == audio.output_rate &&

            bytes_per_sample == audio.output_sample_size )

        return TRUE;

    samples_per_buffer = (driver->sample_rate * 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 = driver->sample_format;

    audio.output_rate = driver->sample_rate;

    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();

    }

    int next_buffer = NEXT_BUFFER();

    result = audio.output_buffers[next_buffer];

    if( result->status == BUFFER_FULL )

        return NULL;

    else {

        audio.write_buffer = next_buffer;

        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];

    if( current->status == BUFFER_FULL ) {

        // Current read buffer has data, which we've just emptied

        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 ) {

            current->posn = 0;

            return current;

        }

        else return NULL;

    } 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 < AUDIO_CHANNEL_COUNT; 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->start))[channel->posn];

                    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;

                                channel->loop = LOOP_LOOPED;

                            } 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->start))[channel->posn] << 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;

                                channel->loop = LOOP_LOOPED;

                            } 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;

                        channel->posn++;

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

                            if( channel->loop ) {

                                channel->posn = channel->loop_start;

                                channel->loop = LOOP_LOOPED;

                                channel->adpcm_predict = 0;

                                channel->adpcm_step = 0;

                            } else {

                                audio_stop_channel(i);

                                j = num_samples;

                                break;

                            }

                        }

                        uint8_t data = ((uint8_t *)(arm_mem + channel->start))[channel->posn>>1];

                        if( channel->posn&1 ) {

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

                        } else {

                            adpcm_yamaha_decode_nibble( channel, data & 0x0F );

                        }

                    }

                }

                break;

            default:

                break;

            }

        }

    }

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

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

    if( buf->status == BUFFER_FULL ) {

        buf = audio_next_write_buffer();

        if( buf == NULL ) { // no available space

            return;

        }

    }

    switch( audio.output_format & AUDIO_FMT_SAMPLE_MASK ) {

    case AUDIO_FMT_FLOAT: {

        float scale = 1.0/SHRT_MAX;

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

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

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

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

            buf->posn += 8;

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

                buf = audio_next_write_buffer();

                if( buf == NULL ) {

                    break;

                }

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

            }

        }

        break;

    }

    case AUDIO_FMT_16BIT: {

        int16_t *data = (int16_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 ) {

                buf = audio_next_write_buffer();

                if( buf == NULL ) {

                    // All buffers are full

                    break;

                }

                data = (int16_t *)&buf->data[0];

            }

        }

        break;

    }

    case AUDIO_FMT_8BIT: {

        int8_t *data = (int8_t *)&buf->data[buf->posn];

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

            *data++ = (int8_t)(result_buf[j][0] >> 16);

            *data++ = (int8_t)(result_buf[j][1] >> 16);	

            buf->posn += 2;

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

                buf = audio_next_write_buffer();

                if( buf == NULL ) {

                    // All buffers are full

                    break;

                }

                buf = audio.output_buffers[audio.write_buffer];

                data = (int8_t *)&buf->data[0];

            }

        }

        break;

    }

    }

}

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

audio_channel_t audio_get_channel( int channel ) 

{

    return &audio.channels[channel];

}

void audio_start_stop_channel( int channel, gboolean start )

{

    if( audio.channels[channel].active ) {

        if( !start ) {

            audio_stop_channel(channel);

        }

    } else if( start ) {

        audio_start_channel(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].active = TRUE;

    if( audio.channels[channel].sample_format == AUDIO_FMT_ADPCM ) {

        audio.channels[channel].adpcm_step = 0;

        audio.channels[channel].adpcm_predict = 0;

        uint8_t data = ((uint8_t *)(arm_mem + audio.channels[channel].start))[0];

        adpcm_yamaha_decode_nibble( &audio.channels[channel], data & 0x0F );

    }

}