/* Here's an OpenAL implementation modeled after 
 * the SDL_SoundMixer which was built ontop of SDL_Mixer
 * and SDL_Sound. 
 * Eric Wing
 */

#include "SDL_ALmixer.h"

#include "SDL.h" /* For SDL_GetTicks(), SDL_Delay */
#include "al.h" /* OpenAL */
#include "alc.h" /* For creating OpenAL contexts */

/* For malloc, bsearch, qsort */
#include <stdlib.h>

/* For memcpy */
#include <string.h>

#if 0
/* for toupper */
#include <ctype.h>
/* for strrchr */
#include <string.h>
#endif

/* Currently used in the output debug functions */
#include <stdio.h>

/* My own CircularQueue implementation needed
 * to work around the Nvidia problem of the
 * lack of a buffer query.
 */
#include "CircularQueue.h"

/* Needed for the Mutex locks (and threads if enabled) */
#include "SDL_thread.h"

/* Because of the API differences between the Loki
 * and Creative distributions, we need to know which
 * version to use. The LOKI distribution currently
 * has AL_BYTE_LOKI defined in altypes.h which
 * I will use as a flag to identify the distributions.
 * If this is ever removed, I might revert back to the
 * if defined(_WIN32) or defined(__APPLE__) test to
 * identify the Creative dist. 
 * I'm not sure if or how the Nvidia distribution differs
 * from the Creative distribution. So for
 * now, the Nvidia distribution gets lumped with the 
 * Creative dist and I hope nothing will break.
 * My aluGetErrorString may be the most vulnerable.
 */
#ifdef AL_BYTE_LOKI
	#define USING_LOKI_AL_DIST
	/* This is a short term fix to get around the 
	 * queuing problem with non-power of two buffer sizes.
	 * Hopefully the maintainers will fix this before 
	 * we're ready to ship.
	 */
	#define ENABLE_LOKI_QUEUE_FIX_HACK

	/* The AL_GAIN in the Loki dist doesn't seem to do
	 * what I want/expect it to do. I want to use it for 
	 * Fading, but it seems to work like an off/on switch.
	 * 0 = off, >0 = on. 
	 * The AL_GAIN_LINEAR_LOKI switch seems to do what 
	 * I want, so I'll redefine it here so the code is consistent
	 */
	/* Update: I've changed the source volume implementations 
	 * to use AL_MAX_GAIN, so I don't think I need this block 
	 * of code anymore. The listener uses AL_GAIN, but I 
	 * hope they got this one right since there isn't a AL_MAX_GAIN
	 * for the listener.
	 */
/*
	#undef AL_GAIN
	#include "alexttypes.h"
	#define AL_GAIN AL_GAIN_LINEAR_LOKI
*/
#else
	/* Might need to run other tests to figure out the DIST */
	/* I've been told that Nvidia doesn't define constants
	 * in the headers like Creative. Instead of
	 * #define AL_REFERENCE_DISTANCE 0x1020, 
	 * Nvidia prefers you query OpenAL for a value.
	 * int AL_REFERENCE_DISTANCE = alGetEnumValue(ALubyte*)"AL_REFERNECE_DISTANCE");
	 * So I'm assuming this means the Nvidia lacks this value.
	 * If this is the case,
	 * I guess we can use it to identify the Nvidia dist
	 */
	#ifdef AL_REFERENCE_DISTANCE
		#define USING_CREATIVE_AL_DIST
	#else
		#define USING_NVIDIA_AL_DIST
	#endif
#endif

#ifdef ENABLE_LOKI_QUEUE_FIX_HACK
/* Need memset to zero out data */
#include <string.h>
#endif


/* Seek issues for predecoded samples:
 * The problem is that OpenAL makes us copy an
 * entire buffer if we want to use it. This 
 * means we potentially have two copies of the 
 * same data. For predecoded data, this can be a 
 * large amount of memory. However, for seek 
 * support, I need to be able to get access to 
 * the original data so I can set byte positions.
 * The following flags let you disable seek support
 * if you don't want the memory hit, keep everything,
 * or let you try to minimize the memory wasted by
 * fetching it from the OpenAL buffer if needed
 * and making a copy of it.
 * Update: I don't think I need this flag anymore. I've made the
 * effects of this user customizable by the access_data flag on load.
 * If set to true, then seek and data callbacks work, with the 
 * cost of more memory and possibly CPU for copying the data through
 * the callbacks. If false, then the extra memory is freed, but 
 * you don't get the features.
 */
/*
#define DISABLE_PREDECODED_SEEK
*/
/* Problem: Even though alGetBufferi(., AL_DATA, .)
 * is in the Creative Programmer's reference,
 * it actually isn't in the dist. (Invalid enum
 * in Creative, can't compile in Loki.)
 * So we have to keep it disabled
 */
#define DISABLE_SEEK_MEMORY_OPTIMIZATION

#ifndef DISABLE_SEEK_MEMORY_OPTIMIZATION
/* Needed for memcpy */
#include <string.h>
#endif

/* Old way of doing things: 
#if defined(_WIN32) || defined(__APPLE__)			
#define USING_CREATIVE_AL_DIST
#else
#define USING_LOKI_AL_DIST
#endif
*/

/************ REMOVE  ME (Don't need anymore) ********/
#if 0 
/* Let's get fancy and see if triple buffering
 * does anything good for us 
 * Must be 2 or more or things will probably break
 */
#define NUMBER_OF_QUEUE_BUFFERS 	5
/* This is the number of buffers that are queued up
 * when play first starts up. This should be at least 1
 * and no more than NUMBER_OF_QUEUE_BUFFERS
 */
#define NUMBER_OF_START_UP_BUFFERS 	2
#endif
/************ END REMOVE  ME (Don't need anymore) ********/

static Uint8 ALmixer_Initialized = 0;
/* This should be set correctly by Init */
static Uint32 ALmixer_Frequency_global = ALMIXER_DEFAULT_FREQUENCY;

/* Will be initialized in Init */
static Sint32 Number_of_Channels_global = 0;
static Sint32 Number_of_Reserve_Channels_global = 0;
static Uint32 Is_Playing_global = 0;

/* This is for a simple lock system. It is not meant to be good,
 * but just sufficient to minimize/avoid threading issues
 */
static SDL_mutex* simple_lock;

#ifdef ENABLE_ALMIXER_THREADS
static SDL_Thread* Stream_Thread_global = NULL;
#endif


/* If ENABLE_PARANOID_SIGNEDNESS_CHECK is used,
 * these values will be reset on Init()
 * Consider these values Read-Only.
 */

#define ALMIXER_SIGNED_VALUE 127
#define ALMIXER_UNSIGNED_VALUE 255

#ifdef ENABLE_PARANOID_SIGNEDNESS_CHECK
static Uint16 SIGN_TYPE_16_BIT_FORMAT = AUDIO_S16SYS;
static Uint16 SIGN_TYPE_8_BIT_FORMAT = AUDIO_S8;
#else
static const Uint16 SIGN_TYPE_16_BIT_FORMAT = AUDIO_S16SYS;
static const Uint16 SIGN_TYPE_8_BIT_FORMAT = AUDIO_S8;
#endif

static struct ALmixer_Channel
{
	Uint8 channel_in_use;
	Uint8 callback_update; /* For streaming determination */
	Uint8 needs_stream; /* For streaming determination */
	Uint8 halted;
	Uint8 paused;
	ALuint alsource;
	ALmixer_Data* almixer_data;
	Sint32 loops;
	Sint32 expire_ticks;
	Uint32 start_time;

	Uint8 fade_enabled;
	Uint32 fade_expire_ticks;
	Uint32 fade_start_time;
	ALfloat fade_inv_time;
	ALfloat fade_start_volume;
	ALfloat fade_end_volume;
	ALfloat max_volume;
	ALfloat min_volume;

	/* Do we need other flags?
	Uint8 *samples;
	int volume;
	int looping;
	int tag;
	Uint32 expire;
	Uint32 start_time;
	Mix_Fading fading;
	int fade_volume;
	Uint32 fade_length;
	Uint32 ticks_fade;
	effect_info *effects;
	*/
} *ALmixer_Channel_List = NULL;

/* This will be used to find a channel if the user supplies a source */
typedef struct Source_Map
{
    ALuint source;
    Sint32 channel;
} Source_Map;
/* Keep an array of all sources with their associated channel */
static Source_Map* Source_Map_List;

static int Compare_Source_Map(const void* a, const void* b)
{
    return ( ((Source_Map*)a)->source - ((Source_Map*)b)->source );
}

/* Sort by channel instead of source */
static int Compare_Source_Map_by_channel(const void* a, const void* b)
{
    return ( ((Source_Map*)a)->channel - ((Source_Map*)b)->channel );
}

/* Compare by albuffer */
static int Compare_Buffer_Map(const void* a, const void* b)
{
    return ( ((Buffer_Map*)a)->albuffer - ((Buffer_Map*)b)->albuffer );
}

/* This is for the user defined callback via 
 * ALmixer_ChannelFinished()
 */
static void (*Channel_Done_Callback)(Sint32 channel, void* userdata) = NULL;
static void* Channel_Done_Callback_Userdata = NULL;
static void (*Channel_Data_Callback)(Sint32 which_channel, Uint8* data, Uint32 num_bytes, Uint32 frequency, Uint8 channels, Uint8 bitdepth, Uint16 format, Uint8 decode_mode) = NULL;


/* I thought OpenAL seemed to lack an error number to string converter...
* but I was wrong. Apparently they call it alGetString() which
* breaks from the OpenGL gluGetErrorString() convention.
* (And since the documentation for OpenAL is so bad, I didn't see
   * it until I had already written my own aluGetErrorString().)
* So for convenience, I will just call alGetString from here.
*/
static const ALubyte* aluGetErrorString(ALenum error)
{
	return alGetString(error);
	
#if 0
	switch(error)
	{
		case AL_NO_ERROR:
			return NULL;
			break;
		case AL_INVALID_NAME:
			return "Invalid name (ID)";
			break;
		case AL_INVALID_VALUE:
			return "Invalid value";
			break;
		case AL_OUT_OF_MEMORY:
			return "Out of memory";
			break;
			/* Damn, even the error values are skewed between distributions */
			/* For the Creative Labs distributions (don't know about Nvidia) */
			/* For the Loki based distributions */
#ifdef USING_LOKI_AL_DIST
		case AL_ILLEGAL_ENUM:
			return "Invalid enum value";
			break;
		case AL_ILLEGAL_COMMAND:
			return "Requested operation is not valid";
			break;
			/*
#elif USING_CREATIVE_AL_DIST
			 */
#else
		case AL_INVALID_ENUM:
			return "Invalid enum value";
			break;
		case AL_INVALID_OPERATION:
			return "Requested operation is not valid";
			break;
			/*
#elif USING_NVIDIA_AL_DIST
			 */
			/*
			 case alGetEnumValue((ALubyte*)"??????");
			 */
#endif
		default:
			return "Unknown error value passed to aluGetErrorString()";
			break;
	}
	
	/* Make compiler happy */
	return NULL;
#endif
}


static void PrintQueueStatus(ALuint source)
{
	ALint buffers_queued = 0;
	ALint buffers_processed = 0;
	ALenum error;
	
	/* Get the number of buffers still queued */
	alGetSourcei(
		source,
		AL_BUFFERS_QUEUED, 
		&buffers_queued
	);
	
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "Error in PrintQueueStatus, Can't get buffers_queued: %s\n",
			aluGetErrorString(error));				
	}
	/* Get the number of buffers processed
	 * so we know if we need to refill 
	 */
	alGetSourcei(
		source,
		AL_BUFFERS_PROCESSED, 
		&buffers_processed
	);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "Error in PrintQueueStatus, Can't get buffers_processed: %s\n",
			aluGetErrorString(error));				
	}
	
	fprintf(stderr, "For source: %d, buffers_queued=%d, buffers_processed=%d\n",
			source,
			buffers_queued,
			buffers_processed);
}



static void Init_Channel(Sint32 channel)
{

fprintf(stderr, "Init channel %d\n", channel);
	ALmixer_Channel_List[channel].channel_in_use = 0;
	ALmixer_Channel_List[channel].callback_update = 0;
	ALmixer_Channel_List[channel].needs_stream = 0;
	ALmixer_Channel_List[channel].paused = 0;
	ALmixer_Channel_List[channel].halted = 0;
	ALmixer_Channel_List[channel].loops = 0;
	
	ALmixer_Channel_List[channel].expire_ticks = 0;
	ALmixer_Channel_List[channel].start_time = 0;

	ALmixer_Channel_List[channel].fade_enabled = 0;
	ALmixer_Channel_List[channel].fade_expire_ticks = 0;
	ALmixer_Channel_List[channel].fade_start_time = 0;
	ALmixer_Channel_List[channel].fade_inv_time = 0.0f;
	ALmixer_Channel_List[channel].fade_start_volume = 0.0f;
	ALmixer_Channel_List[channel].fade_end_volume = 0.0f;
	ALmixer_Channel_List[channel].max_volume = 1.0f;
	ALmixer_Channel_List[channel].min_volume = 0.0f;
	
	ALmixer_Channel_List[channel].almixer_data = NULL;
}
/* Quick helper function to clean up a channel 
 * after it's done playing */
static void Clean_Channel(Sint32 channel)
{
	ALenum error;
	ALmixer_Channel_List[channel].channel_in_use = 0;
	ALmixer_Channel_List[channel].callback_update = 0;
	ALmixer_Channel_List[channel].needs_stream = 0;
	ALmixer_Channel_List[channel].paused = 0;
	ALmixer_Channel_List[channel].halted = 0;
	ALmixer_Channel_List[channel].loops = 0;
	
		
	ALmixer_Channel_List[channel].expire_ticks = 0;
	ALmixer_Channel_List[channel].start_time = 0;

	ALmixer_Channel_List[channel].fade_enabled = 0;
	ALmixer_Channel_List[channel].fade_expire_ticks = 0;
	ALmixer_Channel_List[channel].fade_start_time = 0;
	ALmixer_Channel_List[channel].fade_inv_time = 0.0f;
	ALmixer_Channel_List[channel].fade_start_volume = 0.0f;
	ALmixer_Channel_List[channel].fade_end_volume = 0.0f;

	alSourcef(ALmixer_Channel_List[channel].alsource, AL_MAX_GAIN, 
		ALmixer_Channel_List[channel].max_volume);

	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "10Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n",
			aluGetErrorString(error));				
	}
	
	alSourcef(ALmixer_Channel_List[channel].alsource, AL_MIN_GAIN, 
		ALmixer_Channel_List[channel].min_volume);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "11Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n",
			aluGetErrorString(error));				
	}
	
	if(ALmixer_Channel_List[channel].almixer_data != NULL)
	{
		if(ALmixer_Channel_List[channel].almixer_data->in_use > 0)
		{
			ALmixer_Channel_List[channel].almixer_data->in_use--;
		}
	}
	/* Needed to determine if rewind is needed, can't reset */
	/*
	ALmixer_Channel_List[channel].almixer_data->eof = 0;
	*/

	ALmixer_Channel_List[channel].almixer_data = NULL;
}


#if 0
/* Not needed anymore because not doing any fileext checks.
 *
 * Unfortunately, strcasecmp isn't portable so here's a
 * reimplementation of it (taken from SDL_sound)
 */
static int ALmixer_strcasecmp(const char* x, const char* y)
{
	int ux, uy;
	
	if (x == y)  /* same pointer? Both NULL? */
		return(0);

	if (x == NULL)
		return(-1);

	if (y == NULL)
		return(1);
	   
	do
	{
		ux = toupper((int) *x);
		uy = toupper((int) *y);
		if (ux > uy)
			return(1);
		else if (ux < uy)
			return(-1);
		x++;
		y++;
	} while ((ux) && (uy));

    return(0);
}
#endif

					
/* What shoud this return?
 * 127 for signed, 255 for unsigned
 */
static Uint8 GetSignednessValue(Uint16 format)
{
	switch(format)
	{
		case AUDIO_U8:
		case AUDIO_U16LSB:
		case AUDIO_U16MSB:
			return ALMIXER_UNSIGNED_VALUE;
			break;
		case AUDIO_S8:
		case AUDIO_S16LSB:
		case AUDIO_S16MSB:
			return ALMIXER_SIGNED_VALUE;
			break;
		default:
			return 0;
	}
	return 0;
}


static Uint8 GetBitDepth(Uint16 format)
{
	Uint8 bit_depth = 16;
	
	switch(format)
	{
		case AUDIO_U8:
		case AUDIO_S8:
			bit_depth = 8;
			break;
				
		case AUDIO_U16LSB:
		/*
		case AUDIO_U16:
		*/
		case AUDIO_S16LSB:
		/*
		case AUDIO_S16:
		*/
		case AUDIO_U16MSB:
		case AUDIO_S16MSB:
		/*
		case AUDIO_U16SYS:
		case AUDIO_S16SYS:
		*/
			bit_depth = 16;
			break;
		default:
			bit_depth = 0;
	}
	return bit_depth;
}
	
/* Need to translate between SDL/SDL_Sound audiospec
 * and OpenAL conventions */
static ALenum TranslateFormat(Sound_AudioInfo* info)
{
	Uint8 bit_depth;
	
	bit_depth = GetBitDepth(info->format);
	if(0 == bit_depth)
	{
		fprintf(stderr, "Warning: Unknown bit depth. Setting to 16\n");
		bit_depth = 16;
	}
	
	if(2 == info->channels)
	{
		if(16 == bit_depth)
		{
			return AL_FORMAT_STEREO16;
		}
		else
		{
			return AL_FORMAT_STEREO8;
		}
	}
	else
	{
		if(16 == bit_depth)
		{
			return AL_FORMAT_MONO16;
		}
		else
		{
			return AL_FORMAT_MONO8;
		}
	}
	/* Make compiler happy. Shouldn't get here */
	return AL_FORMAT_STEREO16;
}

/**************** REMOVED ****************************/
/* This was removed because I originally thought
 * OpenAL could return a pointer to the buffer data,
 * but I was wrong. If something like that is ever
 * implemented, then this might become useful.
 */
#if 0
/* Reconstruct_Sound_Sample and Set_AudioInfo only
 * are needed if the Seek memory optimization is 
 * used. Also, the Loki dist doesn't seem to support
 * AL_DATA which I need for it.
 */
#ifndef DISABLE_SEEK_MEMORY_OPTIMIZATION

static void Set_AudioInfo(Sound_AudioInfo* info, ALint frequency, ALint bits, ALint channels)
{
	info->rate = (Uint32)frequency;
	info->channels = (Uint8)channels;
	
	/* Not sure if it should be signed or unsigned. Hopefully
	 * that detail won't be needed.
	 */
	if(8 == bits)
	{
		info->format = AUDIO_U8;
	}
	else
	{
		info->format = AUDIO_U16SYS;
	}
	fprintf(stderr, "Audio info: freq=%d, chan=%d, format=%d\n", 
		info->rate, info->channels, info->format);
	
}


static Sint32 Reconstruct_Sound_Sample(ALmixer_Data* data)
{
	ALenum error;	
	ALint* data_from_albuffer;
	ALint freq;
	ALint bits;
	ALint channels;
	ALint size;
	
	/* Create memory all initiallized to 0. */
	data->sample = (Sound_Sample*)calloc(1, sizeof(Sound_Sample));
	if(NULL == data->sample)
	{
		ALmixer_SetError("Out of memory for Sound_Sample");
		return -1;
	}

	/* Clear errors */
	alGetError();

	alGetBufferi(data->buffer[0], AL_FREQUENCY, &freq);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		ALmixer_SetError("alGetBufferi(AL_FREQUENCT): %s", aluGetErrorString(error) );
		free(data->sample);
		data->sample = NULL;
		return -1;
	}
	
	alGetBufferi(data->buffer[0], AL_BITS, &bits);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		ALmixer_SetError("alGetBufferi(AL_BITS): %s", aluGetErrorString(error) );
		free(data->sample);
		data->sample = NULL;
		return -1;
	}

	alGetBufferi(data->buffer[0], AL_CHANNELS, &channels);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		ALmixer_SetError("alGetBufferi(AL_CHANNELS): %s", aluGetErrorString(error) );
		free(data->sample);
		data->sample = NULL;
		return -1;
	}

	alGetBufferi(data->buffer[0], AL_SIZE, &size);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		ALmixer_SetError("alGetBufferi(AL_SIZE): %s", aluGetErrorString(error) );
		free(data->sample);
		data->sample = NULL;
		return -1;
	}
	
	alGetBufferi(data->buffer[0], AL_DATA, data_from_albuffer);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		ALmixer_SetError("alGetBufferi(AL_DATA): %s", aluGetErrorString(error) );
		free(data->sample);
		data->sample = NULL;
		return -1;
	}
	
	if(size <= 0)
	{
		ALmixer_SetError("No data in al buffer");
		free(data->sample);
		data->sample = NULL;
		return -1;
	}
	
	/* Now that we have all the attributes, we need to 
	 * allocate memory for the buffer and reconstruct
	 * the AudioInfo attributes.
	 */
	data->sample->buffer = malloc(size*sizeof(Uint8));
	if(NULL == data->sample->buffer)
	{
		ALmixer_SetError("Out of memory for sample->buffer");
		free(data->sample);
		data->sample = NULL;
		return -1;
	}
	
	memcpy(data->sample->buffer, data_from_albuffer, size);
	data->sample->buffer_size = size;

	/* Fill up the Sound_AudioInfo structures */
	Set_AudioInfo(&data->sample->desired, freq, bits, channels);
	Set_AudioInfo(&data->sample->actual, freq, bits, channels);

	return 0;
}
	
#endif /* End DISABLE_SEEK_MEMORY_OPTIMIZATION */
#endif
/*************** END REMOVED *************************/	

static void Invoke_Channel_Done_Callback(Sint32 channel)
{
	if(NULL == Channel_Done_Callback)
	{
		return;
	}
	Channel_Done_Callback(channel, Channel_Done_Callback_Userdata);
}

static Sint32 LookUpBuffer(ALuint buffer, Buffer_Map* buffer_map_list, Uint32 num_items_in_list)
{
	/* Only the first value is used for the key */
	Buffer_Map key = { 0, 0, NULL, 0 };
	Buffer_Map* found_item = NULL;
	key.albuffer = buffer;

	/* Use the ANSI C binary search feature (yea!) */
	found_item = (Buffer_Map*)bsearch(&key, buffer_map_list, num_items_in_list, sizeof(Buffer_Map), Compare_Buffer_Map);
	if(NULL == found_item)
	{
		ALmixer_SetError("Can't find buffer");
		return -1;
	}
	return found_item->index;
}


/* FIXME: Need to pass back additional info to be useful.
 * Bit rate, stereo/mono (num chans), time in msec?
 * Precoded/streamed flag so user can plan for future data?
 */
static void Invoke_Channel_Data_Callback(Sint32 which_channel, Uint8* data, Uint32 num_bytes, Uint32 frequency, Uint8 channels, Uint16 format, Uint8 decode_mode)
{
/*
	fprintf(stderr, "%x %x %x %x, bytes=%d, whichchan=%d, freq=%d, channels=%d\n", data[0], data[1], data[2], data[3], num_bytes, channels, frequency, channels);
*/
	if(NULL == Channel_Data_Callback)
	{
		return;
	}
	Channel_Data_Callback(which_channel, data, num_bytes, frequency, channels, GetBitDepth(format), format, decode_mode);
}

static void Invoke_Predecoded_Channel_Data_Callback(Sint32 channel, ALmixer_Data* data)
{
	if(NULL == data->sample)
	{
		return;
	}
	/* The buffer position is complicated because if the current data was seeked,
	 * we must adjust the buffer to the seek position
	 */
	Invoke_Channel_Data_Callback(channel, 
		(((Uint8*) data->sample->buffer) + (data->total_bytes - data->loaded_bytes) ),
		data->loaded_bytes,
		data->sample->desired.rate,
		data->sample->desired.channels,
		data->sample->desired.format,
		ALMIXER_DECODE_ALL
	);
}

static void Invoke_Streamed_Channel_Data_Callback(Sint32 channel, ALmixer_Data* data, ALuint buffer)
{
	Sint32 index;
	if(NULL == data->buffer_map_list)
	{
		return;
	}
	index = LookUpBuffer(buffer, data->buffer_map_list, data->max_queue_buffers);
	/* This should catch the case where all buffers are unqueued
	 * and the "current" buffer is id: 0
	 */
	if(-1 == index)
	{
		return;
	}
	Invoke_Channel_Data_Callback(channel, 
		data->buffer_map_list[index].data,
		data->buffer_map_list[index].num_bytes,
		data->sample->desired.rate,
		data->sample->desired.channels,
		data->sample->desired.format,
		ALMIXER_DECODE_STREAM
	);
}

/* From SDL_Sound's playsound. Converts milliseconds to byte positions.
 * This is needed for seeking on predecoded samples 
 */
static Uint32 Convert_Msec_To_Byte_Pos(Sound_AudioInfo *info, Uint32 ms)
{
	float frames_per_ms;
	Uint32 frame_offset;
	Uint32 frame_size;
	fprintf(stderr, "In convert\n" );
	if(info == NULL)
	{
		fprintf(stderr, "Error, info is NULL\n");
	}
	else
	{
		fprintf(stderr, "Not an error: info is not NULL\n");
	}
	fprintf(stderr, "The rate=%d\n", info->rate);

	/* "frames" == "sample frames" */
	frames_per_ms = ((float) info->rate) / 1000.0f;
	fprintf(stderr, "%f\n", frames_per_ms);
	frame_offset = (Uint32) (frames_per_ms * ((float) ms));
	fprintf(stderr, "%d\n", frame_offset);
	frame_size = (Uint32) ((info->format & 0xFF) / 8) * info->channels;
	fprintf(stderr, "%d\n", frame_size);
	return(frame_offset * frame_size);
} /* cvtMsToBytePos */

static Sint32 Set_Predecoded_Seek_Position(ALmixer_Data* data, Uint32 byte_position)
{
	ALenum error;
	/* clear error */
	alGetError();
	
	/* Is it greater than,  or greater-than or equal to ?? */
	if(byte_position > data->total_bytes)
	{
		/* We can't go past the end, so set to end? */
	fprintf(stderr, "Error, can't seek past end\n");
	
	/* In case the below thing doesn't work, 
	 * just rewind the whole thing.
	 *
		alBufferData(data->buffer[0],
			TranslateFormat(&data->sample->desired), 
			 (Uint8*) data->sample->buffer,
			 data->total_bytes,
			data->sample->desired.rate
		);
	*/
	
		/* I was trying to set to the end, (1 byte remaining),
		 * but I was getting freezes. I'm thinking it might be
		 * another Power of 2 bug in the Loki dist. I tried 2,
		 * and it still hung. 4 didn't hang, but I got a clip
		 * artifact. 8 seemed to work okay.
		 */
		alBufferData(data->buffer[0],
			TranslateFormat(&data->sample->desired), 
			 (((Uint8*) data->sample->buffer) + (data->total_bytes - 8) ),
			 8,
			data->sample->desired.rate
		);
		if( (error = alGetError()) != AL_NO_ERROR)
		{
			ALmixer_SetError("Can't seek past end and alBufferData failed: %s\n", aluGetErrorString(error));
			return -1;
		}
		/* Need to set the loaded_bytes field because I don't trust the OpenAL
		 * query command to work because I don't know if it will mutilate the
		 * size for its own purposes or return the original size
		 */
		 data->loaded_bytes = 8;

		/* Not sure if this should be an error or not */
/*
		ALmixer_SetError("Can't Seek past end");	
		return -1;
*/
		return 0;
	}
	
	alBufferData(data->buffer[0],
		TranslateFormat(&data->sample->desired), 
		&(((Uint8*)data->sample->buffer)[byte_position]),
		data->total_bytes - byte_position,
		data->sample->desired.rate
	);
	if( (error = alGetError()) != AL_NO_ERROR)
	{
		ALmixer_SetError("alBufferData failed: %s\n", aluGetErrorString(error));
		return -1;
	}
	/* Need to set the loaded_bytes field because I don't trust the OpenAL
	 * query command to work because I don't know if it will mutilate the
	 * size for its own purposes or return the original size
	 */
	 data->loaded_bytes = data->total_bytes - byte_position;

	return 0;
}

/* This will compute the total playing time
 * based upon the number of bytes and audio info.
 * (In prinicple, it should compute the time for any given length) 
 */
static Uint32 Compute_Total_Time(Sound_AudioInfo *info, Uint32 total_bytes)
{
	Uint32 bytes_per_sec;
	Uint32 bytes_per_sample;
	double total_sec;
	Uint32 total_msec;
	
	if(0 == total_bytes)
	{
		return 0;
	}
	/* SDL has a mask trick I was not aware of. Mask the upper bits
	 * of the format, and you get 8 or 16 which is the bits per sample.
	 * Divide by 8bits_per_bytes and you get bytes_per_sample
	 */
	bytes_per_sample = (Uint32) ((info->format & 0xFF) / 8);
	/* To compute Bytes per second, do
	 * samples_per_sec * bytes_per_sample * number_of_channels
	 */
	bytes_per_sec = info->rate * bytes_per_sample * info->channels;

	/* Now to get total time (sec), do
	 * total_bytes / bytes_per_sec
	 */
	total_sec = total_bytes / (double)bytes_per_sec;
	
	/* Now convert seconds to milliseconds
	 * Add .5 to the float to do rounding before the final cast
	 */
	total_msec = (Uint32) ( (total_sec * 1000) + 0.5 );
	
	fprintf(stderr, "%d\n", total_msec);
	return total_msec;
} /* End Compute_Total_Time */




/* Because we have multiple queue buffers and OpenAL won't let
 * us access them, we need to keep copies of each buffer around
 */
static Sint32 CopyDataToAccessBuffer(ALmixer_Data* data, Uint32 num_bytes, ALuint buffer)
{
	Sint32 index;
	/* We only want to copy if access_data is true.
	 * This is determined by whether memory has been
	 * allocated in the buffer_map_list or not
	 */
	if(NULL == data->buffer_map_list)
	{
		return -1;
	}
	index = LookUpBuffer(buffer, data->buffer_map_list, data->max_queue_buffers);
	if(-1 == index)
	{
fprintf(stderr, ">>>>>>>CopyData catch, albuffer=%d\n",buffer);
		return -1;
	}
	/* Copy the data to the access buffer */
	memcpy(data->buffer_map_list[index].data, data->sample->buffer, num_bytes);
	data->buffer_map_list[index].num_bytes = data->sample->buffer_size;
	
	return 0;
}


/* For streamed data, gets more data
 * and prepares it in the active Mix_chunk
 */
static Uint32 GetMoreData(ALmixer_Data* data, ALuint buffer)
{
	Uint32 bytes_decoded;
	ALenum error;
	if(NULL == data)
	{
		ALmixer_SetError("Cannot GetMoreData() because ALmixer_Data* is NULL\n");
		return 0;
	}

	bytes_decoded = Sound_Decode(data->sample);
	if(data->sample->flags & SOUND_SAMPLEFLAG_ERROR)
	{
fprintf(stderr, "Sound_Decode triggered an ERROR>>>>>>\n");
		ALmixer_SetError(Sound_GetError());
		/* Force cleanup through FreeData
		Sound_FreeSample(data->sample);
		*/
		return 0;
	}
	
	
	/* Don't forget to add check for EOF */
	/* Will return 0 bytes and pass the buck to check sample->flags */
	if(0 == bytes_decoded)
	{
#if 1 
fprintf(stderr, "Hit eof while trying to buffer\n");
		data->eof = 1;
		if(data->sample->flags & SOUND_SAMPLEFLAG_EOF)
		{
			fprintf(stderr, "\tEOF flag\n");
		}
		if(data->sample->flags & SOUND_SAMPLEFLAG_CANSEEK)
		{
			fprintf(stderr, "\tCanSeek flag\n");
		}
		if(data->sample->flags & SOUND_SAMPLEFLAG_EAGAIN)
		{
			fprintf(stderr, "\tEAGAIN flag\n");
		}
		if(data->sample->flags & SOUND_SAMPLEFLAG_NONE)
		{
			fprintf(stderr, "\tNONE flag\n");
		}
#endif
		return 0;
	}

#ifdef ENABLE_LOKI_QUEUE_FIX_HACK
/******* REMOVE ME ********************************/
/***************** ANOTHER EXPERIEMENT *******************/
	/* The PROBLEM: It seems that the Loki distribution has problems
	 * with Queuing when the buffer size is not a power of two
	 * and additional buffers must come after it.
	 * The behavior is inconsistent, but one of several things
	 * usually happens:
	 *   Playback is normal
	 *   Playback immediately stops after the non-pow2 buffer
	 *   Playback gets distorted on the non-pow2 buffer
	 *   The entire program segfaults.
	 * The workaround is to always specify a power of two buffer size
	 * and hope that SDL_sound always fill it. (By lucky coincidence,
	 * I already submitted the Ogg fix.) However, this won't catch
	 * cases where a loop happens because the read at the end of the
	 * file is typically less than the buffer size.
	 *
	 * This fix addresses this issue, however it may break in
	 * other conditions. Always decode in buffer sizes of powers of 2.
	 * 
	 * The HACK:
	 * If the buffer is short, try filling it up with 0's
 	 * to meet the user requested buffer_size which 
	 * is probably a nice number OpenAL likes, in 
	 * hopes to avoid a possible Loki bug with
	 * short buffers. If looping (which is the main
	 * reason for this), the negative side effect is
	 * that it may take longer for the loop to start
	 * because it must play dead silence. Or if the decoder
	 * doesn't guarantee to return the requested bytes
	 * (like Ogg), then you will get breakup in between
	 * packets.
	 */
	if( (bytes_decoded) < data->sample->buffer_size)
	{
		Uint8 bit_depth;
		Uint8 signedness_value;
		int silence_value;
		/* Crap, memset value needs to be the "silent" value, 
		 * but it will differ for signed/unsigned and bit depth
		 */
		bit_depth = GetBitDepth(data->sample->desired.format);
		signedness_value = GetSignednessValue(data->sample->desired.format);
		if(ALMIXER_SIGNED_VALUE == signedness_value)
		{
			/* I'm guessing that if it's signed, then 0 is the
			 * "silent" value */
			silence_value = 0;
		}
		else
		{
			if(8 == bit_depth)
			{
				/* If 8 bit, I'm guessing it's (2^7)-1 = 127 */ 
				silence_value = 127;
			}
			else
			{
				/* For 16 bit, I'm guessing it's (2^15)-1 = 32767 */
				silence_value = 32767;
			}
		}
		/* Now fill up the rest of the data buffer with the 
		 * silence_value.
		 * I don't think I have to worry about endian issues for
		 * this part since the data is for internal use only
		 * at this point.
		 */
		memset( &( ((Uint8*)(data->sample->buffer))[bytes_decoded] ), silence_value, data->sample->buffer_size - bytes_decoded);
		/* Now reset the bytes_decoded to reflect the entire 
		 * buffer to tell alBufferData what our full size is.
		 */
	fprintf(stderr, "ALTERED bytes decoded for silence: Original end was %d\n", bytes_decoded);
		bytes_decoded = data->sample->buffer_size;
	}
/*********** END EXPERIMENT ******************************/
/******* END REMOVE ME ********************************/
#endif

	/* Now copy the data to the OpenAL buffer */
	/* We can't just set a pointer because the API needs
	 * its own copy to assist hardware acceleration */
	alBufferData(buffer,
		TranslateFormat(&data->sample->desired), 
		data->sample->buffer,
		bytes_decoded,
		data->sample->desired.rate
	);
	if( (error = alGetError()) != AL_NO_ERROR)
	{
		ALmixer_SetError("alBufferData failed: %s\n", aluGetErrorString(error));
		return 0;
	}

	/* If we need to, copy the data also to the access area 
	 * (the function will do the check for us)
	 */
	CopyDataToAccessBuffer(data, bytes_decoded, buffer);
	return bytes_decoded;
}




/********************  EXPERIEMENT **************************** 
 * Test function to force maximum buffer filling during loops
 * REMOVE LATER
 *********************************************/
#if 0
static Sint32 GetMoreData2(ALmixer_Data* data, ALuint buffer)
{
	Sint32 bytes_decoded;
	ALenum error;
	if(NULL == data)
	{
		ALmixer_SetError("Cannot GetMoreData() because ALmixer_Data* is NULL\n");
		return -1;
	}

if(AL_FALSE == alIsBuffer(buffer))
{
	fprintf(stderr, "NOT A BUFFER>>>>>>>>>>>>>>>\n");
	return -1;
}
fprintf(stderr, "Entered GetMoreData222222: buffer id is %d\n", buffer);
	
/*
fprintf(stderr, "Decode in GetMoreData\n");
*/

#if 0
if(buffer%2 == 1)
{
	fprintf(stderr, "Setting buffer size to 16384\n");
	Sound_SetBufferSize(data->sample, 16384);
}
else
{
	fprintf(stderr, "Setting buffer size to 8192\n");
	Sound_SetBufferSize(data->sample, 8192);
}
#endif

	bytes_decoded = Sound_Decode(data->sample);
	if(data->sample->flags & SOUND_SAMPLEFLAG_ERROR)
	{
fprintf(stderr, "Sound_Decode triggered an ERROR>>>>>>\n");
		ALmixer_SetError(Sound_GetError());
		/*
		Sound_FreeSample(data->sample);
		*/
		return -1;
	}
	/* Don't forget to add check for EOF */
	/* Will return 0 bytes and pass the buck to check sample->flags */
	if(0 == bytes_decoded)
	{
#if 1 
fprintf(stderr, "Hit eof while trying to buffer\n");
		data->eof = 1;
		if(data->sample->flags & SOUND_SAMPLEFLAG_EOF)
		{
			fprintf(stderr, "\tEOF flag\n");
		}
		if(data->sample->flags & SOUND_SAMPLEFLAG_CANSEEK)
		{
			fprintf(stderr, "\tCanSeek flag\n");
		}
		if(data->sample->flags & SOUND_SAMPLEFLAG_EAGAIN)
		{
			fprintf(stderr, "\tEAGAIN flag\n");
		}
		if(data->sample->flags & SOUND_SAMPLEFLAG_NONE)
		{
			fprintf(stderr, "\tNONE flag\n");
		}
#endif
		return 0;
	}

	if(bytes_decoded < 16384)
	{
		char* tempbuffer1 = (char*)malloc(16384);
		char* tempbuffer2 = (char*)malloc(16384);
		int retval;
		memcpy(tempbuffer1, data->sample->buffer, bytes_decoded);
		retval = Sound_SetBufferSize(data->sample, 16384-bytes_decoded);
		if(retval == 1)
		{
			Uint32 new_bytes;
			Sound_Rewind(data->sample);
			new_bytes = Sound_Decode(data->sample);
			fprintf(stderr, "Orig bytes: %d, Make up bytes_decoded=%d, total=%d\n", bytes_decoded, new_bytes, new_bytes+bytes_decoded);

			memcpy(tempbuffer2, data->sample->buffer, new_bytes);
			
		retval = Sound_SetBufferSize(data->sample, 16384);
		fprintf(stderr, "Finished reset...now danger copy\n");
			memcpy(data->sample->buffer, tempbuffer1,bytes_decoded);

		fprintf(stderr, "Finished reset...now danger copy2\n");
			memcpy( &( ((char*)(data->sample->buffer))[bytes_decoded] ), tempbuffer2, new_bytes);
			
		fprintf(stderr, "Finished \n");
			
			free(tempbuffer1);
			free(tempbuffer2);
			bytes_decoded += new_bytes;
			fprintf(stderr, "ASSERT bytes should equal 16384: %d\n", bytes_decoded);
		}
		else
		{
			fprintf(stderr, "Experiment failed: %s\n", Sound_GetError());
		}
	}
		
	/* Now copy the data to the OpenAL buffer */
	/* We can't just set a pointer because the API needs
	 * its own copy to assist hardware acceleration */
	alBufferData(buffer,
		TranslateFormat(&data->sample->desired), 
		data->sample->buffer,
		bytes_decoded,
		data->sample->desired.rate
	);
	if( (error = alGetError()) != AL_NO_ERROR)
	{
		ALmixer_SetError("alBufferData failed: %s\n", aluGetErrorString(error));
		return -1;
	}
	
fprintf(stderr, "GetMoreData2222 returning %d bytes decoded\n", bytes_decoded);
	return bytes_decoded;
}
#endif

/************ END EXPERIEMENT - REMOVE ME *************************/









/* This function will look up the source for the corresponding channel */
/* Must return 0 on error instead of -1 because of unsigned int */
static ALuint Internal_GetSource(Sint32 channel)
{
	Sint32 i;
	/* Make sure channel is in bounds */
	if(channel >= Number_of_Channels_global)
	{
		ALmixer_SetError("Requested channel (%d) exceeds maximum channel (%d) because only %d channels are allocated", channel, Number_of_Channels_global-1, Number_of_Channels_global);
		return 0;	
	}
	/* If the user specified -1, then return the an available source */
	if(channel < 0)
	{
		for(i=Number_of_Reserve_Channels_global; i<Number_of_Channels_global; i++)
		{
			if( ! ALmixer_Channel_List[i].channel_in_use )
			{
				return ALmixer_Channel_List[i].alsource;
			}
		}
		/* If we get here, all sources are in use */			
		/* Error message seems too harsh
		ALmixer_SetError("All sources are in use");
		*/
		return 0;
	}
	/* Last case: Return the source for the channel */	
	return ALmixer_Channel_List[channel].alsource; 
}

/* This function will look up the channel for the corresponding source */
static Sint32 Internal_GetChannel(ALuint source)
{
	Sint32 i;
	/* Only the first value is used for the key */
	Source_Map key = { 0, 0 };
	Source_Map* found_item = NULL;
	key.source = source;

	/* If the source is 0, look up the first available channel */
	if(0 == source)
	{
		for(i=Number_of_Reserve_Channels_global; i<Number_of_Channels_global; i++)
		{
			if( ! ALmixer_Channel_List[i].channel_in_use )
			{
				return i;
			}
		}
		/* If we get here, all sources are in use */			
		/* Error message seems too harsh
		ALmixer_SetError("All channels are in use");
		*/
		return -1;
	}
	
	
	/* Else, look up the source and return the channel */
	if(AL_FALSE == alIsSource(source))
	{
		ALmixer_SetError("Is not a source");
		return -1;
	}
	
	/* Use the ANSI C binary search feature (yea!) */
	found_item = (Source_Map*)bsearch(&key, Source_Map_List, Number_of_Channels_global, sizeof(Source_Map), Compare_Source_Map);
	if(NULL == found_item)
	{
		ALmixer_SetError("Source is valid but not registered with ALmixer (to a channel)");
		return -1;
	}
	return found_item->channel;
}



/* This function will find the first available channel (not in use)
 * from the specified start channel. Reserved channels to not qualify
 * as available.
 */
static Sint32 Internal_FindFreeChannel(Sint32 start_channel)
{
	/* Start at the number of reserved so we skip over
	 * all the reserved channels.
	 */
	Sint32 i = Number_of_Reserve_Channels_global;
	/* Quick check to see if we're out of bounds */
	if(start_channel >= Number_of_Channels_global)
	{
		return -1;
	}
	
	/* If the start channel is even higher than the reserved,
	 * then start at the higher value.
	 */
	if(start_channel > Number_of_Reserve_Channels_global)
	{
		i = start_channel;
	}
	
	/* i has already been set */
	for( ; i<Number_of_Channels_global; i++)
	{
		if( ! ALmixer_Channel_List[i].channel_in_use )
		{
			return i;
		}
	}
	/* If we get here, all sources are in use */			
	return -1;
}




/* Note: Behaves, almost like SDL_mixer, but keep in mind
 * that there is no "music" channel anymore, so 0
 * will remove everything. (Note, I no longer allow 0
 * so it gets set to the default number.)
 * Also, callbacks for deleted channels will not be called.
 * I really need to do error checking, for realloc and 
 * GenSources, but reversing the damage is too painful
 * for me to think about at the moment, so it's not in here.
 */
static Sint32 Internal_AllocateChannels(Sint32 numchans)
{
	ALenum error;
	int i;
	/* Return info */
	if(numchans < 0)
	{
		return Number_of_Channels_global;
	}
	if(0 == numchans)
	{
		numchans = ALMIXER_DEFAULT_NUM_CHANNELS;
	}
	/* No change */
	if(numchans == Number_of_Channels_global)
	{
		return Number_of_Channels_global;
	}
	/* We need to increase the number of channels */
	if(numchans > Number_of_Channels_global)
	{
		/* Not sure how safe this is, but SDL_mixer does it
		 * the same way */
		ALmixer_Channel_List = (struct ALmixer_Channel*) realloc( ALmixer_Channel_List, numchans * sizeof(struct ALmixer_Channel));

		/* Allocate memory for the list of sources that map to the channels */
		Source_Map_List = (Source_Map*) realloc(Source_Map_List, numchans * sizeof(Source_Map));

		for(i=Number_of_Channels_global; i<numchans; i++)
		{
			Init_Channel(i);
			/* Generate a new source and associate it with the channel */
			alGenSources(1, &ALmixer_Channel_List[i].alsource);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "12Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n",
			aluGetErrorString(error));				
	}
			/* Copy the source so the SourceMap has it too */
			Source_Map_List[i].source = ALmixer_Channel_List[i].alsource;
			Source_Map_List[i].channel = i;
			/* Clean the channel because there are some things that need to 
			 * be done that can't happen until the source is set
			 */
			Clean_Channel(i);
		}

		/* The Source_Map_List must be sorted by source for binary searches
		 */
		qsort(Source_Map_List, numchans, sizeof(Source_Map), Compare_Source_Map);
	
		Number_of_Channels_global = numchans;
		return numchans;
	}
	/* Need to remove channels. This might be dangerous */
	if(numchans < Number_of_Channels_global)
	{
		for(i=numchans; i<Number_of_Channels_global; i++)
		{
			/* Halt the channel */
			ALmixer_HaltChannel(i);

			/* Delete source associated with the channel */
			alDeleteSources(1, &ALmixer_Channel_List[i].alsource);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "13Testing error: %s\n",
			aluGetErrorString(error));				
	}
		}


		/* Not sure how safe this is, but SDL_mixer does it
		 * the same way */
		ALmixer_Channel_List = (struct ALmixer_Channel*) realloc( ALmixer_Channel_List, numchans * sizeof(struct ALmixer_Channel));

		/* The tricky part is that we must remove the entries
		 * in the source map that correspond to the deleted channels.
		 * We'll resort the map by channels so we can pick them off
		 * in order.
		 */
		qsort(Source_Map_List, Number_of_Channels_global, sizeof(Source_Map), Compare_Source_Map_by_channel);

		/* Deallocate memory for the list of sources that map to the channels */
		Source_Map_List = (Source_Map*) realloc(Source_Map_List, numchans * sizeof(Source_Map));

		/* Now resort the map by source and the correct num of chans */
		qsort(Source_Map_List, numchans, sizeof(Source_Map), Compare_Source_Map);

		/* Reset the number of channels */
		Number_of_Channels_global = numchans;
		return numchans;
	}
	/* Shouldn't ever reach here */
	return -1;
	
}
	
static Sint32 Internal_ReserveChannels(Sint32 num)
{
	/* Can't reserve more than the max num of channels */
	/* Actually, I'll allow it for people who just want to
	 * set the value really high to effectively disable
	 * auto-assignment
	 */

	/* Return the current number of reserved channels */
	if(num < 0)
	{
		return Number_of_Reserve_Channels_global;
	}
	Number_of_Reserve_Channels_global = num;
	return Number_of_Reserve_Channels_global;
}
	




static Sint32 Internal_PlayChannelTimed(Sint32 channel, ALmixer_Data* data, Sint32 loops, Sint32 ticks)
{
	ALenum error;
	int ret_flag = 0;
	if(NULL == data)
	{
		ALmixer_SetError("Can't play because data is NULL\n");
		return -1;
	}

	/* There isn't a good way to share streamed files because
	 * the decoded data doesn't stick around. 
	 * You must "Load" a brand new instance of
	 * the data. If you try using the same data,
	 * bad things may happen. This check will attempt
	 * to prevent sharing
	 */
	if(0 == data->decoded_all)
	{
		if(data->in_use)
		{
			ALmixer_SetError("Can't play shared streamed sample because it is already in use");
			return -1;
		}

		/* Make sure SDL_sound sample is not at EOF.
		 * This mainly affects streamed files,
	 	 * so the check is placed here
		 */
		if(data->eof)
		{
			if( -1 == ALmixer_RewindData(data) )
			{
				ALmixer_SetError("Can't play sample because it is at EOF and cannot rewind");
				return -1;
			}
		}
	}
	/* We need to provide the user with the first available channel */
	if(-1 == channel)
	{
		Sint32 i;
		for(i=Number_of_Reserve_Channels_global; i<Number_of_Channels_global; i++)
		{
			if(0 == ALmixer_Channel_List[i].channel_in_use)
			{
				channel = i;
				break;
			}
		}
		/* if we couldn't find a channel, return an error */
		if(i == Number_of_Channels_global)
		{
			ALmixer_SetError("No channels available for playing");
			return -1;
		}
	}
	/* If we didn't assign the channel number, make sure it's not
	 * out of bounds or in use */
	else
	{
		if(channel >= Number_of_Channels_global)
		{
			ALmixer_SetError("Requested channel (%d) exceeds maximum channel (%d) because only %d channels are allocated", channel, Number_of_Channels_global-1, Number_of_Channels_global);
			return -1;
		}
		else if(ALmixer_Channel_List[channel].channel_in_use)
		{
			ALmixer_SetError("Requested channel (%d) is in use", channel, Number_of_Channels_global-1, Number_of_Channels_global);
			return -1;
		}	
	}
	/* Make sure the user doesn't enter some meaningless value */
	if(loops < -1)
	{
		loops = -1;
	}
							
	/* loops will probably have to change to be controlled by SDL_Sound */

	/* Set up the initial values for playing */
	ALmixer_Channel_List[channel].channel_in_use = 1;
	data->in_use++;
		
	/* Shouldn't need updating until a callback is fired
	 * (assuming that we call Play in this function 
	 */
	ALmixer_Channel_List[channel].needs_stream = 0;
	ALmixer_Channel_List[channel].almixer_data = data;
	ALmixer_Channel_List[channel].start_time = SDL_GetTicks();

	/* If user entered -1 (or less), set to -1 */
	if(ticks < 0)
	{
		ALmixer_Channel_List[channel].expire_ticks = -1;
	}
	else
	{
		ALmixer_Channel_List[channel].expire_ticks = ticks;
	}

	
	ALmixer_Channel_List[channel].halted = 0;
	ALmixer_Channel_List[channel].paused = 0;

	/* Ran just use OpenAL to control loops if predecoded and infinite */
	ALmixer_Channel_List[channel].loops = loops; 
	if( (-1 == loops) && (data->decoded_all) )
	{
		alSourcei(ALmixer_Channel_List[channel].alsource, AL_LOOPING, AL_TRUE);
	}
	else
	{
		alSourcei(ALmixer_Channel_List[channel].alsource, AL_LOOPING, AL_FALSE);
	}
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "13Testing error: %s\n",
			aluGetErrorString(error));				
	}

#if 0
	/* Because of the corner case, predecoded
	 * files must add +1 to the loops.
	 * Streams do not have this problem
	 * because they can use the eof flag to 
	 * avoid the conflict.
	 * Sharing data chunks prevents the use of the eof flag.
	 * Since streams, cannot share, only predecoded
	 * files are affected 
	 */
	if(data->decoded_all)
	{
		/* Corner Case: Now that play calls are pushed
		 * off to update(), the start call must
		 * also come through here. So, start loops
		 * must be +1 
		 */
		if(-1 == loops)
		{
			/* -1 is a special case, and you don't want
			 * to add +1 to it */
			ALmixer_Channel_List[channel].loops = -1;
			alSourcei(ALmixer_Channel_List[channel].alsource, AL_LOOPING, AL_TRUE);
		}
		else
		{
			ALmixer_Channel_List[channel].loops = loops+1;
			alSourcei(ALmixer_Channel_List[channel].alsource, AL_LOOPING, AL_FALSE);
		}
	}
	else
	{
		ALmixer_Channel_List[channel].loops = loops;
		/* Can we really loop on streamed data? */
		alSourcei(ALmixer_Channel_List[channel].alsource, AL_LOOPING, AL_TRUE);
	}
#endif

	/* Should I start playing here or pass the buck to update? */
	/* Unlike SDL_SoundMixer, I think I'll do it here because
	 * this library isn't a *total* hack and OpenAL has more 
	 * built in functionality I need, so less needs to be 
	 * controlled and directed through the update function.
	 * The downside is less functionality is centralized.
	 * The upside is that the update function should be
	 * easier to maintain.
	 */

	/* Clear the error flag */
	alGetError();
	if(data->decoded_all)
	{
		/* Bind the data to the source */
		alSourcei(
			ALmixer_Channel_List[channel].alsource, 
			AL_BUFFER, 
			data->buffer[0]);
		if((error = alGetError()) != AL_NO_ERROR)
		{
			ALmixer_SetError("Could not bind data to source: %s",
				aluGetErrorString(error) );
			Clean_Channel(channel);
			return -1;
		}

		/* Make data available if access_data is enabled */
		Invoke_Predecoded_Channel_Data_Callback(channel, data);
	}
	else
	{
		/* Need to use the streaming buffer for binding */

		Uint32 bytes_returned;
		Uint32 j;
		data->num_buffers_in_use=0;
/****** MODIFICATION must go here *********/
		/* Since buffer queuing is pushed off until here to 
		 * avoid buffer conflicts, we must start reading 
		 * data here. First we make sure we have at least one
		 * packet. Then we queue up until we hit our limit.
		 */
		bytes_returned = GetMoreData(
			data,
			data->buffer[0]);
		if(0 == bytes_returned)
		{
			/* No data or error */
			ALmixer_SetError("Could not get data for streamed PlayChannel: %s", ALmixer_GetError());
			Clean_Channel(channel);
			return -1;
		}
		/* Increment the number of buffers in use */
		data->num_buffers_in_use++;
		

		/* Now we need to fill up the rest of the buffers.
		 * There is a corner case where we run out of data
		 * before the last buffer is filled.
		 * Stop conditions are we run out of 
		 * data or we max out our preload buffers.
		 */
			
	fprintf(stderr, "Filling buffer #%d (AL id is %d)\n", 0, data->buffer[0]);
		for(j=1; j<data->num_startup_buffers; j++)
		{
	fprintf(stderr, "Filling buffer #%d (AL id is %d)\n", j, data->buffer[j]);
	/*
	fprintf(stderr, ">>>>>>>>>>>>>>>>>>HACK for GetMoreData2\n");
	*/
		bytes_returned = GetMoreData(
				data,
				data->buffer[j]);
		/* 
		 * This might be a problem. I made a mistake with the types. I accidentally
		 * made the bytes returned an Sint32 and returned -1 on error.
		 * Bytes returned should be a Uint32, so now I no longer have a -1 case
		 * to check. I hope I didn't break anything here
		 */
		#if 0
			if(bytes_returned < 0)
			{
				/* Error found */
				ALmixer_SetError("Could not get data for additional startup buffers for PlayChannel: %s", ALmixer_GetError());
				/* We'll continue on because we do have some valid data */
				ret_flag = -1;
				break;
			}
			else if(0 == bytes_returned)
		#endif
			if(0 == bytes_returned)
			{
				/* No more data to buffer */
				/* Check for loops */
				if( ALmixer_Channel_List[channel].loops != 0 )
				{
fprintf(stderr, "Need to rewind. In RAMPUP, handling loop\n");
					if(0 == Sound_Rewind(data->sample))
					{
fprintf(stderr, "error in rewind\n"); 
						ALmixer_SetError( Sound_GetError() );
						ALmixer_Channel_List[channel].loops = 0;
						ret_flag = -1;
						/* We'll continue on because we do have some valid data */
						break;
					}
					/* Remember to reset the data->eof flag */
					data->eof = 0;
					if(ALmixer_Channel_List[channel].loops > 0)
					{
						ALmixer_Channel_List[channel].loops--;
fprintf(stderr, "Inside 000 >>>>>>>>>>Loops=%d\n", ALmixer_Channel_List[channel].loops); 
					}
					/* Would like to redo the loop, but due to 
					 * Sound_Rewind() bugs, we would risk falling 
					 * into an infinite loop
					 */
					bytes_returned = GetMoreData(
						data,
						data->buffer[j]);
					if(bytes_returned <= 0)
					{
						ALmixer_SetError("Could not get data: %s", ALmixer_GetError());
						/* We'll continue on because we do have some valid data */
						ret_flag = -1;
						break;
					}
				}
				else
				{
					/* No loops to do so quit here */
					break;
				}
			}
			/* Increment the number of buffers in use */
			data->num_buffers_in_use++;
		}
		/*
	fprintf(stderr, "In PlayChannel, about to queue: source=%d, num_buffers_in_use=%d\n",
			ALmixer_Channel_List[channel].alsource, 
			data->num_buffers_in_use);
*/
		
		alSourceQueueBuffers(
			ALmixer_Channel_List[channel].alsource, 
			data->num_buffers_in_use, 
			data->buffer);
		if((error = alGetError()) != AL_NO_ERROR)
		{
			ALmixer_SetError("Could not bind data to source: %s",
				aluGetErrorString(error) );
			Clean_Channel(channel);
			return -1;
		}
		/* This is part of the hideous Nvidia workaround. In order to figure out
		 * which buffer to show during callbacks (for things like
		 * o-scopes), I must keep a copy of the buffers that are queued in my own
		 * data structure. This code will be called only if
		 * "access_data" was set, indicated by whether the queue is NULL.
		 */
		if(data->circular_buffer_queue != NULL)
		{
			Uint32 k;
			Uint32 queue_ret_flag;
			for(k=0; k<data->num_buffers_in_use; k++)
			{
				queue_ret_flag = CircularQueueUnsignedInt_PushBack(data->circular_buffer_queue, data->buffer[k]);
				if(0 == queue_ret_flag)
				{
					fprintf(stderr, "Serious internal error: CircularQueue could not push into queue.\n");
					ALmixer_SetError("Serious internal error: CircularQueue failed to push into queue");
				}
				/*
				else
				{
					fprintf(stderr, "Queue in PlayTimed\n");
					CircularQueueUnsignedInt_Print(data->circular_buffer_queue);
				}
				 */
			}
		}
				
		
/****** END **********/
	}
	/* We have finished loading the data (predecoded or queued)
	 * so now we can play 
	 */
	alSourcePlay(ALmixer_Channel_List[channel].alsource);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		ALmixer_SetError("Play failed: %s",
			aluGetErrorString(error) );
		Clean_Channel(channel);
		return -1;
	}

	/* Add to the counter that something is playing */
	Is_Playing_global++;
	if(-1 == ret_flag)
	{
			fprintf(stderr, "BACKDOOR ERROR >>>>>>>>>>>>>>>>>>\n");
		return -1;
	}
	return channel;
}


/* In case the user wants to specify a source instead of a channel,
 * they may use this function. This function will look up the 
 * source-to-channel map, and convert the call into a
 * PlayChannelTimed() function call.
 * Returns the channel it's being played on.
 * Note: If you are prefer this method, then you need to be careful
 * about using PlayChannel, particularly if you request the
 * first available channels because source and channels have 
 * a one-to-one mapping in this API. It is quite easy for 
 * a channel/source to already be in use because of this.
 * In this event, an error message will be returned to you.
 */
static ALuint Internal_PlaySourceTimed(ALuint source, ALmixer_Data* data, Sint32 loops, Sint32 ticks)
{
	Sint32 channel;
	Sint32 retval;
	if(0 == source)
	{
		retval = Internal_PlayChannelTimed(-1, data, loops, ticks);
		if(-1 == retval)
		{
			return 0;
		}
		else
		{
			return Internal_GetSource(retval);
		}
	}
	
	channel = Internal_GetChannel(source);
	if(-1 == channel)
	{
		ALmixer_SetError("Cannot Play source: %s", ALmixer_GetError());
		return 0;
	}
	retval = Internal_PlayChannelTimed(channel, data, loops, ticks);
	if(-1 == retval)
	{
		return 0;
	}
	else
	{
		return source;
	}
	/* make compiler happy */
	return 0;
}



/* Will return the number of channels halted
 * or 0 for error
 */
static Sint32 Internal_HaltChannel(Sint32 channel)
{
	Sint32 retval = 0;
	Sint32 counter = 0;
	ALenum error;
	ALint buffers_still_queued;
	ALint buffers_processed;

	if(channel >= Number_of_Channels_global)
	{
		ALmixer_SetError("Cannot halt channel %d because it exceeds maximum number of channels (%d)\n", channel, Number_of_Channels_global);
		return -1;
	}
	/* If the user specified a specific channel */
	if(channel >= 0)
	{
		fprintf(stderr, "Halt on channel %d\n", channel);
		/* only need to process channel if in use */
		if(ALmixer_Channel_List[channel].channel_in_use)
		{
			alSourceStop(ALmixer_Channel_List[channel].alsource);
			if((error = alGetError()) != AL_NO_ERROR)
			{
				fprintf(stderr, "14Testing error: %s\n",
					aluGetErrorString(error));				
			}
			/* Here's the situation. My old method of using
			 * alSourceUnqueueBuffers() seemed to be invalid in light
			 * of all the problems I suffered through with getting 
			 * the CoreData backend to work with this code.
			 * As such, I'm changing all the code to set the buffer to
			 * AL_NONE. Furthermore, the queued vs. non-queued issue
			 * doesn't need to apply here. For non-queued, Loki,
			 * Creative Windows, and CoreAudio seem to leave the 
			 * buffer queued (Old Mac didn't.) For queued, we need to 
			 * remove the processed buffers and force remove the
			 * still-queued buffers.
			 */
			fprintf(stderr, "Halt on channel %d, channel in use\n", channel);
			alGetSourcei(
				ALmixer_Channel_List[channel].alsource,
				AL_BUFFERS_QUEUED, &buffers_still_queued
			);
			if((error = alGetError()) != AL_NO_ERROR)
			{
				fprintf(stderr, "17Testing Error with buffers_still_queued: %s",
					aluGetErrorString(error));
				ALmixer_SetError("Failed detecting still queued buffers: %s",
					aluGetErrorString(error) );
				retval = -1;
			}
			alGetSourcei(
				ALmixer_Channel_List[channel].alsource,
				AL_BUFFERS_PROCESSED, &buffers_processed
			);
			if((error = alGetError()) != AL_NO_ERROR)
			{
				fprintf(stderr, "17Testing Error with buffers_processed: %s",
					aluGetErrorString(error));
				ALmixer_SetError("Failed detecting still processed buffers: %s",
					aluGetErrorString(error) );
				retval = -1;
			}
			/* If either of these is greater than 0, it means we need
			 * to clear the source
			 */
			if((buffers_still_queued > 0) || (buffers_processed > 0))
			{
				alSourcei(ALmixer_Channel_List[channel].alsource,
					AL_BUFFER,
					AL_NONE);
				if((error = alGetError()) != AL_NO_ERROR)
				{
					fprintf(stderr, "17Testing Error with clearing buffer from source: %s",
						aluGetErrorString(error));
					ALmixer_SetError("Failed to clear buffer from source: %s",
						aluGetErrorString(error) );
					retval = -1;
				}
			}

			ALmixer_Channel_List[channel].almixer_data->num_buffers_in_use  = 0;

			Clean_Channel(channel);
			Is_Playing_global--;
			/* Launch callback for consistency? */
			Invoke_Channel_Done_Callback(channel);
			counter++;
		}
	}
	/* The user wants to halt all channels */
	else
	{
		Sint32 i;
		for(i=0; i<Number_of_Channels_global; i++)
		{
			fprintf(stderr, "Halting channel %d\n", i);
			fprintf(stderr, "in use %d\n", ALmixer_Channel_List[i].channel_in_use );
			/* only need to process channel if in use */
			if(ALmixer_Channel_List[i].channel_in_use)
			{
				fprintf(stderr, "SourceStop %d\n", i);
				alSourceStop(ALmixer_Channel_List[i].alsource);
				if((error = alGetError()) != AL_NO_ERROR)
				{
					fprintf(stderr, "19Testing error: %s\n",
						aluGetErrorString(error));				
				}

				/* Here's the situation. My old method of using
				 * alSourceUnqueueBuffers() seemed to be invalid in light
				 * of all the problems I suffered through with getting 
				 * the CoreData backend to work with this code.
				 * As such, I'm changing all the code to set the buffer to
				 * AL_NONE. Furthermore, the queued vs. non-queued issue
				 * doesn't need to apply here. For non-queued, Loki,
				 * Creative Windows, and CoreAudio seem to leave the 
				 * buffer queued (Old Mac didn't.) For queued, we need to 
				 * remove the processed buffers and force remove the
				 * still-queued buffers.
				 */
				fprintf(stderr, "Halt on channel %d, channel in use\n", channel);
				alGetSourcei(
					ALmixer_Channel_List[i].alsource,
					AL_BUFFERS_QUEUED, &buffers_still_queued
				);
				if((error = alGetError()) != AL_NO_ERROR)
				{
					fprintf(stderr, "17Testing Error with buffers_still_queued: %s",
						aluGetErrorString(error));
					ALmixer_SetError("Failed detecting still queued buffers: %s",
						aluGetErrorString(error) );
					retval = -1;
				}
				alGetSourcei(
					ALmixer_Channel_List[i].alsource,
					AL_BUFFERS_PROCESSED, &buffers_processed
				);
				if((error = alGetError()) != AL_NO_ERROR)
				{
					fprintf(stderr, "17Testing Error with buffers_processed: %s",
						aluGetErrorString(error));
					ALmixer_SetError("Failed detecting still processed buffers: %s",
						aluGetErrorString(error) );
					retval = -1;
				}
				/* If either of these is greater than 0, it means we need
				 * to clear the source
				 */
				if((buffers_still_queued > 0) || (buffers_processed > 0))
				{
					alSourcei(ALmixer_Channel_List[i].alsource,
						AL_BUFFER,
						AL_NONE);
					if((error = alGetError()) != AL_NO_ERROR)
					{
						fprintf(stderr, "17Testing Error with clearing buffer from source: %s",
							aluGetErrorString(error));
						ALmixer_SetError("Failed to clear buffer from source: %s",
							aluGetErrorString(error) );
						retval = -1;
					}
				}
				
				ALmixer_Channel_List[i].almixer_data->num_buffers_in_use  = 0;

	fprintf(stderr, "Clean channel %d\n", i);
				Clean_Channel(i);
				Is_Playing_global--;
				/* Launch callback for consistency? */
	fprintf(stderr, "Callback%d\n", i);
				Invoke_Channel_Done_Callback(i);

				/* Increment the counter */
				counter++;
			}
			/* Let's halt everything just in case there 
			 * are bugs.
			 */
			/*
			else
			{
				alSourceStop(ALmixer_Channel_List[channel].alsource);
				/ * Can't clean because the in_use counter for
				 * data will get messed up * /
				Clean_Channel(channel);
			}
			*/
			/* Just in case */
			Is_Playing_global = 0;
		}
	}
	if(-1 == retval)
	{		
		return -1;	
	}
	return counter;
}


/* Will return the source halted or the total number of channels
 * if all were halted or 0 for error
 */
static Sint32 Internal_HaltSource(ALuint source)
{
	Sint32 channel;
	if(0 == source)
	{
		/* Will return the number of sources halted */
		return Internal_HaltChannel(-1);
	}
	
	channel = Internal_GetChannel(source);
	if(-1 == channel)
	{
		ALmixer_SetError("Cannot halt source: %s", ALmixer_GetError());
		return 0;
	}
	return Internal_HaltChannel(channel);
}


/* This will rewind the SDL_Sound sample for streamed
 * samples and start buffering up the data for the next
 * playback. This may require samples to be halted
 */
static Sint32 Internal_RewindData(ALmixer_Data* data)
{
	Sint32 retval = 0;
	/*
	Sint32 bytes_returned;
	Sint32 i;
	*/
	if(NULL == data)
	{
		ALmixer_SetError("Cannot rewind because data is NULL\n");
		return -1;
	}


	/* Might have to require Halt */
	/* Okay, we assume Halt or natural stop has already
	 * cleared the data buffers
	 */
	if(data->in_use)
	{
		fprintf(stderr, "Warning sample is in use. May not be able to rewind\n");
		/*
		ALmixer_SetError("Data is in use. Cannot rewind unless all sources using the data are halted\n");
		return -1;
		*/
	}
		
	
	/* Because Seek can alter things even in predecoded data,
	 * decoded data must also be rewound 
	 */
	if(data->decoded_all)
	{
		data->eof = 0;

#if 0
#if defined(DISABLE_PREDECODED_SEEK)
		/* Since we can't seek predecoded stuff, it should be rewound */
		return 0;
#elif !defined(DISABLE_SEEK_MEMORY_OPTIMIZATION)
		/* This case is if the Sound_Sample has been deleted.
		 * It assumes the data is already at the beginning.
		 */
		if(NULL == data->sample)
		{
			return 0;
		}
		/* Else, the sample has already been reallocated,
		 * and we can fall to normal behavior
		 */
#endif
#endif
		/* If access_data, was enabled, the sound sample 
		 * still exists and we can do stuff. 
		 * If it's NULL, we can't do anything, but
		 * it should already be "rewound".
		 */
		if(NULL == data->sample)
		{
			return 0;
		}
		/* Else, the sample has already been reallocated,
		 * and we can fall to normal behavior
		 */

		Set_Predecoded_Seek_Position(data, 0);
	/*
		return data->total_bytes;
	*/
		return 0;
	}
	
	/* Remaining stuff for streamed data */
	
fprintf(stderr, "Rewinding for stream\n");
	data->eof = 0;
	retval = Sound_Rewind(data->sample);
	if(0 == retval)
	{
		ALmixer_SetError( Sound_GetError() );
		return -1;
	}
fprintf(stderr, "Rewinding succeeded\n");
fprintf(stderr, "calling GetMoreData for Rewinding for stream\n");
#if 0
	/* Clear error */
	alGetError();
	for(i=0; i<data->num_buffers; i++)
	{
		bytes_returned = GetMoreData(data, data->buffer[i]);
		if(-1 == bytes_returned)
		{
			return -1;
		}
		else if(0 == bytes_returned)
		{
			return -1;
		}
		retval += bytes_returned;
		
	}
#endif

	
fprintf(stderr, "end Rewinding for stream\n");
	
	return retval;
}




static Sint32 Internal_RewindChannel(Sint32 channel)
{
	Sint32 retval = 0;
	ALenum error;
	ALint state;

	if(channel >= Number_of_Channels_global)
	{
		ALmixer_SetError("Cannot rewind channel %d because it exceeds maximum number of channels (%d)\n", channel, Number_of_Channels_global);
		return -1;
	}

	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "24Testing error: %s\n",
			aluGetErrorString(error));				
	}
	/* Clear error */
	alGetError();
	
	/* If the user specified a specific channel */
	if(channel >= 0)
	{
		/* only need to process channel if in use */
		if(ALmixer_Channel_List[channel].channel_in_use)
		{

			/* What should I do? Do I just rewind the channel
			 * or also rewind the data? Since the data is
			 * shared, let's make it the user's responsibility
			 * to rewind the data.
			 */
			if(ALmixer_Channel_List[channel].almixer_data->decoded_all)
			{
				alGetSourcei(
					ALmixer_Channel_List[channel].alsource,
					AL_SOURCE_STATE, &state
				);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "25Testing error: %s\n",
			aluGetErrorString(error));				
	}
				alSourceRewind(ALmixer_Channel_List[channel].alsource);
				if((error = alGetError()) != AL_NO_ERROR)
				{
					ALmixer_SetError("%s",
						aluGetErrorString(error) );
					retval = -1;
				}
				/* Need to resume playback if it was originally playing */
				if(AL_PLAYING == state)
				{
					alSourcePlay(ALmixer_Channel_List[channel].alsource);
					if((error = alGetError()) != AL_NO_ERROR)
					{
						ALmixer_SetError("%s",
							aluGetErrorString(error) );
						retval = -1;
					}
				}
				else if(AL_PAUSED == state)
				{
					/* HACK: The problem is that when paused, after
					 * the Rewind, I can't get it off the INITIAL
					 * state without restarting
					 */
					alSourcePlay(ALmixer_Channel_List[channel].alsource);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "25Testing error: %s\n",
			aluGetErrorString(error));				
	}
					alSourcePause(ALmixer_Channel_List[channel].alsource);
					if((error = alGetError()) != AL_NO_ERROR)
					{
						ALmixer_SetError("%s",
							aluGetErrorString(error) );
						retval = -1;
					}
				}
			}
			else
			{
				/* Streamed data is different. Rewinding the channel
				 * does no good. Rewinding the data will have an
				 * effect, but it will be lagged based on how
				 * much data is queued. Recommend users call Halt
				 * before rewind if they want immediate results.
				 */
				retval = Internal_RewindData(ALmixer_Channel_List[channel].almixer_data);
			}
		}
	}
	/* The user wants to halt all channels */
	else
	{
		Sint32 i;
		for(i=0; i<Number_of_Channels_global; i++)
		{
	fprintf(stderr, "in use %d\n", ALmixer_Channel_List[i].channel_in_use );
			/* only need to process channel if in use */
			if(ALmixer_Channel_List[i].channel_in_use)
			{
				/* What should I do? Do I just rewind the channel
				 * or also rewind the data? Since the data is
				 * shared, let's make it the user's responsibility
				 * to rewind the data.
				 */
				if(ALmixer_Channel_List[i].almixer_data->decoded_all)
				{
					alGetSourcei(
						ALmixer_Channel_List[i].alsource,
						AL_SOURCE_STATE, &state
					);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "26Testing error: %s\n",
			aluGetErrorString(error));				
	}
					alSourceRewind(ALmixer_Channel_List[i].alsource);
					if((error = alGetError()) != AL_NO_ERROR)
					{
						ALmixer_SetError("%s",
							aluGetErrorString(error) );
						retval = -1;
					}
					/* Need to resume playback if it was originally playing */
					if(AL_PLAYING == state)
					{
						alSourcePlay(ALmixer_Channel_List[i].alsource);
						if((error = alGetError()) != AL_NO_ERROR)
						{
							ALmixer_SetError("%s",
								aluGetErrorString(error) );
							retval = -1;
						}
					}
					else if(AL_PAUSED == state)
					{
						/* HACK: The problem is that when paused, after
						 * the Rewind, I can't get it off the INITIAL
						 * state without restarting
						 */
						alSourcePlay(ALmixer_Channel_List[i].alsource);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "27Testing error: %s\n",
			aluGetErrorString(error));				
	}
						alSourcePause(ALmixer_Channel_List[i].alsource);
						if((error = alGetError()) != AL_NO_ERROR)
						{
							ALmixer_SetError("%s",
								aluGetErrorString(error) );
							retval = -1;
						}
					}
				}
				else
				{
					/* Streamed data is different. Rewinding the channel
					 * does no good. Rewinding the data will have an
					 * effect, but it will be lagged based on how
					 * much data is queued. Recommend users call Halt
					 * before rewind if they want immediate results.
					 */
					retval = Internal_RewindData(ALmixer_Channel_List[i].almixer_data);
				}
			}
		}
	}
	return retval;
}


static Sint32 Internal_RewindSource(ALuint source)
{
	Sint32 channel;
	if(0 == source)
	{
		return Internal_RewindChannel(-1);
	}
	
	channel = Internal_GetChannel(source);
	if(-1 == channel)
	{
		ALmixer_SetError("Cannot rewind source: %s", ALmixer_GetError());
		return 0;
	}
	return Internal_RewindChannel(channel);
}



/* Returns the channel or number of channels actually paused */

static Sint32 Internal_PauseChannel(Sint32 channel)
{
	ALenum error;
	ALint state;
	Sint32 retval = 0;
	Sint32 counter = 0;
	
	if(channel >= Number_of_Channels_global)
	{
		ALmixer_SetError("Cannot pause channel %d because it exceeds maximum number of channels (%d)\n", channel, Number_of_Channels_global);
		return -1;
	}

	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "28Testing error: %s\n",
			aluGetErrorString(error));				
	}
	/* Clear error */
	alGetError();
	
	/* If the user specified a specific channel */
	if(channel >= 0)
	{
		fprintf(stderr, "Pause on channel %d\n", channel);
		/* only need to process channel if in use */
		if(ALmixer_Channel_List[channel].channel_in_use)
		{
			/* We don't want to repause if already
			 * paused because the fadeout/expire
			 * timing will get messed up
			 */
			alGetSourcei(
				ALmixer_Channel_List[channel].alsource,
				AL_SOURCE_STATE, &state
			);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "29Testing error: %s\n",
			aluGetErrorString(error));				
	}
			if(AL_PLAYING == state)
			{
				/* Count the actual number of channels being paused */
				counter++;
				
				alSourcePause(ALmixer_Channel_List[channel].alsource);
				if((error = alGetError()) != AL_NO_ERROR)
				{
					ALmixer_SetError("%s",
						aluGetErrorString(error) );
					retval = -1;
				}
				/* We need to pause the expire time count down */
				if(ALmixer_Channel_List[channel].expire_ticks != -1)
				{
					Uint32 current_time = SDL_GetTicks();
					Uint32 diff_time;
					diff_time = current_time - 
						ALmixer_Channel_List[channel].start_time;
					/* When we unpause, we will want to reset
					 * the start time so we can continue
					 * to base calculations off GetTicks().
					 * This means we need to subtract the amount
					 * of time already used up from expire_ticks.
					 */
					ALmixer_Channel_List[channel].expire_ticks =
						ALmixer_Channel_List[channel].expire_ticks -
						diff_time;
					/* Because -1 is a special value, we can't 
					 * allow the time to go negative
					 */
					if(ALmixer_Channel_List[channel].expire_ticks < 0)
					{
						ALmixer_Channel_List[channel].expire_ticks = 0;
					}
				}
				/* Do the same as expire time for fading */
				if(ALmixer_Channel_List[channel].fade_enabled)
				{
					Uint32 current_time = SDL_GetTicks();
					Uint32 diff_time;
					diff_time = current_time - 
						ALmixer_Channel_List[channel].fade_start_time;
					/* When we unpause, we will want to reset
					 * the start time so we can continue
					 * to base calculations off GetTicks().
					 * This means we need to subtract the amount
					 * of time already used up from expire_ticks.
					 */
					ALmixer_Channel_List[channel].fade_expire_ticks =
						ALmixer_Channel_List[channel].fade_expire_ticks -
						diff_time;
					/* Don't allow the time to go negative */
					if(ALmixer_Channel_List[channel].expire_ticks < 0)
					{
						ALmixer_Channel_List[channel].expire_ticks = 0;
					}
				} /* End fade check */
			} /* End if PLAYING */
		} /* End If in use */
	} /* End specific channel */
	/* The user wants to halt all channels */
	else
	{
		Sint32 i;
		for(i=0; i<Number_of_Channels_global; i++)
		{
	fprintf(stderr, "Pausing channel %d\n", i);
	fprintf(stderr, "in use %d\n", ALmixer_Channel_List[i].channel_in_use );
			/* only need to process channel if in use */
			if(ALmixer_Channel_List[i].channel_in_use)
			{
				/* We don't want to repause if already
				 * paused because the fadeout/expire
				 * timing will get messed up
				 */
				alGetSourcei(
					ALmixer_Channel_List[i].alsource,
					AL_SOURCE_STATE, &state
				);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "30Testing error: %s\n",
			aluGetErrorString(error));				
	}
				if(AL_PLAYING == state)
				{
					/* Count the actual number of channels being paused */
					counter++;
						
						fprintf(stderr, "SourcePause %d\n", i);
					alSourcePause(ALmixer_Channel_List[i].alsource);
					if((error = alGetError()) != AL_NO_ERROR)
					{
						ALmixer_SetError("%s",
							aluGetErrorString(error) );
						retval = -1;
					}
					/* We need to pause the expire time count down */
					if(ALmixer_Channel_List[i].expire_ticks != -1)
					{
						Uint32 current_time = SDL_GetTicks();
						Uint32 diff_time;
						diff_time = current_time - 
							ALmixer_Channel_List[i].start_time;
						/* When we unpause, we will want to reset
						 * the start time so we can continue
						 * to base calculations off GetTicks().
						 * This means we need to subtract the amount
						 * of time already used up from expire_ticks.
						 */
						ALmixer_Channel_List[i].expire_ticks =
							ALmixer_Channel_List[i].expire_ticks -
							diff_time;
						/* Because -1 is a special value, we can't 
						 * allow the time to go negative
						 */
						if(ALmixer_Channel_List[i].expire_ticks < 0)
						{
							ALmixer_Channel_List[i].expire_ticks = 0;
						}
					}
					/* Do the same as expire time for fading */
					if(ALmixer_Channel_List[i].fade_enabled)
					{
						Uint32 current_time = SDL_GetTicks();
						Uint32 diff_time;
						diff_time = current_time - 
							ALmixer_Channel_List[i].fade_start_time;
						/* When we unpause, we will want to reset
						 * the start time so we can continue
						 * to base calculations off GetTicks().
						 * This means we need to subtract the amount
						 * of time already used up from expire_ticks.
						 */
						ALmixer_Channel_List[i].fade_expire_ticks =
							ALmixer_Channel_List[i].fade_expire_ticks -
							diff_time;
						/* Don't allow the time to go negative */
						if(ALmixer_Channel_List[i].expire_ticks < 0)
						{
							ALmixer_Channel_List[i].expire_ticks = 0;
						}
					} /* End fade check */	
				} /* End if PLAYING */
			} /* End channel in use */
		} /* End for-loop */
	}
	if(-1 == retval)
	{
		return -1;
	}
	return counter;	
}

/* Returns the channel or number of channels actually paused */
static Sint32 Internal_PauseSource(ALuint source)
{
	Sint32 channel;
	if(0 == source)
	{
		return Internal_PauseChannel(-1);
	}
	
	channel = Internal_GetChannel(source);
	if(-1 == channel)
	{
		ALmixer_SetError("Cannot pause source: %s", ALmixer_GetError());
		return -1;
	}
	return Internal_PauseChannel(channel);
}



static Sint32 Internal_ResumeChannel(Sint32 channel)
{
	ALint state;
	ALenum error;
	Sint32 retval = 0;
	Sint32 counter = 0;
	
	if(channel >= Number_of_Channels_global)
	{
		ALmixer_SetError("Cannot pause channel %d because it exceeds maximum number of channels (%d)\n", channel, Number_of_Channels_global);
		return -1;
	}

	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "31Testing error: %s\n",
			aluGetErrorString(error));				
	}
	/* Clear error */
	alGetError();

	/* If the user specified a specific channel */
	if(channel >= 0)
	{
		fprintf(stderr, "Pause on channel %d\n", channel);
		/* only need to process channel if in use */
		if(ALmixer_Channel_List[channel].channel_in_use)
		{
			alGetSourcei(
				ALmixer_Channel_List[channel].alsource,
				AL_SOURCE_STATE, &state
			);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "32Testing error: %s\n",
			aluGetErrorString(error));				
	}
			if(AL_PAUSED == state)
			{
				/* Count the actual number of channels resumed */
				counter++;

				/* We need to resume the expire time count down */
				if(ALmixer_Channel_List[channel].expire_ticks != -1)
				{
					ALmixer_Channel_List[channel].start_time = SDL_GetTicks();
				}
				/* Do the same as expire time for fading */
				if(ALmixer_Channel_List[channel].fade_enabled)
				{
					ALmixer_Channel_List[channel].fade_start_time = SDL_GetTicks();
				}	

				alSourcePlay(ALmixer_Channel_List[channel].alsource);
				if((error = alGetError()) != AL_NO_ERROR)
				{
					ALmixer_SetError("%s",
						aluGetErrorString(error) );
					retval = -1;
				}
			}
		fprintf(stderr, "Pause on channel %d, channel in use\n", channel);
		}
	}
	/* The user wants to halt all channels */
	else
	{
		Sint32 i;
		for(i=0; i<Number_of_Channels_global; i++)
		{
	fprintf(stderr, "Pausing channel %d\n", i);
	fprintf(stderr, "in use %d\n", ALmixer_Channel_List[i].channel_in_use );
			/* only need to process channel if in use */
			if(ALmixer_Channel_List[i].channel_in_use)
			{
	fprintf(stderr, "SourcePause %d\n", i);
				alGetSourcei(
					ALmixer_Channel_List[i].alsource,
					AL_SOURCE_STATE, &state
				);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "33Testing error: %s\n",
			aluGetErrorString(error));				
	}
				if(AL_PAUSED == state)
				{
					/* Count the actual number of channels resumed */
					counter++;

					/* We need to resume the expire time count down */
					if(ALmixer_Channel_List[i].expire_ticks != -1)
					{
						ALmixer_Channel_List[i].start_time = SDL_GetTicks();
					}
					/* Do the same as expire time for fading */
					if(ALmixer_Channel_List[i].fade_enabled)
					{
						ALmixer_Channel_List[i].fade_start_time = SDL_GetTicks();
					}	
						
					alSourcePlay(ALmixer_Channel_List[i].alsource);
					if((error = alGetError()) != AL_NO_ERROR)
					{
						ALmixer_SetError("%s",
							aluGetErrorString(error) );
						retval = -1;
					}
				}
			}
		}
	}
	if(-1 == retval)
	{
		return -1;
	}
	return counter;	
}


static Sint32 Internal_ResumeSource(ALuint source)
{
	Sint32 channel;
	if(0 == source)
	{
		return Internal_ResumeChannel(-1);
	}
	
	channel = Internal_GetChannel(source);
	if(-1 == channel)
	{
		ALmixer_SetError("Cannot resume source: %s", ALmixer_GetError());
		return -1;
	}
	return Internal_ResumeChannel(channel);
}


/* Might consider setting eof to 0 as a "feature"
 * This will allow seek to end to stay there because
 * Play automatically rewinds if at the end */
static Sint32 Internal_Seek(ALmixer_Data* data, Uint32 msec)
{
	Sint32 retval;
	
	if(NULL == data)
	{
		ALmixer_SetError("Cannot Seek because data is NULL");
		return -1;
	}
	
	/* Seek for predecoded files involves moving the chunk pointer around */
	if(data->decoded_all)
	{
		Uint32 byte_position;

		/* OpenAL doesn't seem to like it if I change the buffer
		 * while playing (crashes), so I must require that Seek only
		 * be done when the data is not in use. 
		 * Since data may be shared among multiple sources,
		 * I can't shut them down myself, so I have to return an error.
		 */
		if(data->in_use)
		{
			ALmixer_SetError("Cannot seek on predecoded data while instances are playing");
			return -1;
		}
#if 0
#if defined(DISABLE_PREDECODED_SEEK)
		ALmixer_SetError("Seek support for predecoded samples was not compiled in");
		return -1;

#elif !defined(DISABLE_SEEK_MEMORY_OPTIMIZATION)
		/* By default, ALmixer frees the Sound_Sample for predecoded
		 * samples because of the potential memory waste.
		 * However, to seek a sample, we need to have a full
		 * copy of the data around. So the strategy is to
		 * recreate a hackish Sound_Sample to be used for seeking
		 * purposes. If Sound_Sample is NULL, we will reallocate
		 * memory for it and then procede as if everything 
		 * was normal.
		 */
		if(NULL == data->sample)
		{
			if( -1 == Reconstruct_Sound_Sample(data) )
			{
				return -1;
			}
		}
#endif
#endif
		/* If access_data was set, then we still have the 
		 * Sound_Sample and we can move around in the data.
		 * If it was not set, the data has been freed and we 
		 * cannot do anything because there is no way to 
		 * recover the data because OpenAL won't let us
		 * get access to the buffers
		 */
		if(NULL == data->sample)
		{
			ALmixer_SetError("Cannot seek because access_data flag was set false when data was initialized");
			return -1;
		}
		
		fprintf(stderr, "Calling convert\n");
		byte_position = Convert_Msec_To_Byte_Pos(&data->sample->desired, msec);
		fprintf(stderr, "Calling Set_Predecoded_Seek...%d\n", byte_position);
		return( Set_Predecoded_Seek_Position(data, byte_position) );
	}
	else
	{
		/* Reset eof flag?? */
		data->eof = 0;
		retval = Sound_Seek(data->sample, msec);
		if(0 == retval)
		{
			ALmixer_SetError(Sound_GetError());
			
		fprintf(stderr, "Sound seek error: %s\n", ALmixer_GetError());
			/* Try rewinding to clean up? */
/*
			Internal_RewindData(data);
*/
			return -1;
		}
		return 0;
	}

	return 0;
}			
		


static Sint32 Internal_FadeInChannelTimed(Sint32 channel, ALmixer_Data* data, Sint32 loops, Uint32 fade_ticks, Sint32 expire_ticks)
{
	ALfloat value;
	ALenum error;
	ALfloat original_value;
	Uint32 current_time = SDL_GetTicks();
	Sint32 retval;

	
	
	if(channel >= Number_of_Channels_global)
	{
			ALmixer_SetError("Requested channel (%d) exceeds maximum channel (%d) because only %d channels are allocated", channel, Number_of_Channels_global-1, Number_of_Channels_global);
		return -1;
	}
	/* Let's call PlayChannelTimed to do the job. 
	 * There are two catches:
	 * First is that we must set the volumes before the play call(s).
	 * Second is that we must initialize the channel values
	 */

	if(channel < 0)
	{
		/* This might cause a problem for threads/race conditions.
		 * We need to set the volume on an unknown channel,
		 * so we need to request a channel first. Remember
		 * that requesting a channel doesn't lock and it 
		 * could be surrendered to somebody else before we claim it.
		 */
		channel = Internal_GetChannel(0);
		if(-1 == channel)
		{
			return -1;
		}
	}	
	else if(ALmixer_Channel_List[channel].channel_in_use)
	{
		ALmixer_SetError("Channel %d is already in use", channel);
		return -1;
	}

	
	/* Get the original volume in case of a problem */
	alGetSourcef(ALmixer_Channel_List[channel].alsource,
		AL_MAX_GAIN, &original_value);
	
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "35Testing error: %s\n",
			aluGetErrorString(error));				
	}
	/* Get the Max volume */
	/*
	alGetSourcef(ALmixer_Channel_List[channel].alsource,
		AL_MAX_GAIN, &value);
	ALmixer_Channel_List[channel].fade_end_volume = value;
	*/
	ALmixer_Channel_List[channel].fade_end_volume = 
		ALmixer_Channel_List[channel].max_volume;

	fprintf(stderr, "MAX gain: %f\n", value);
	/* Get the Min volume */
	alGetSourcef(ALmixer_Channel_List[channel].alsource,
		AL_MIN_GAIN, &value);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "36Testing error: %s\n",
			aluGetErrorString(error));				
	}
	ALmixer_Channel_List[channel].fade_start_volume = value;
	fprintf(stderr, "MIN gain: %f\n", value);
	
	/* Set the actual volume */
	alSourcef(ALmixer_Channel_List[channel].alsource,
		AL_MAX_GAIN, value);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "37Testing error: %s\n",
			aluGetErrorString(error));				
	}
	
	
	/* Now call PlayChannelTimed */
	retval = Internal_PlayChannelTimed(channel, data, loops, expire_ticks);
	if(-1 == retval)
	{
		/* Chance of failure is actually pretty high since 
		 * a channel might already be in use or streamed
		 * data can be shared
		 */
		/* Restore the original value to avoid accidental 
		 * distruption of playback
		 */
		alSourcef(ALmixer_Channel_List[channel].alsource,
			AL_MAX_GAIN, original_value);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "38Testing error: %s\n",
			aluGetErrorString(error));				
	}
		return retval;
	}

	/* We can't accept 0 as a value because of div-by-zero.
	 * If zero, just call PlayChannelTimed at normal
	 * volume
	 */
	if(0 == fade_ticks)
	{
		alSourcef(ALmixer_Channel_List[channel].alsource,
			AL_MAX_GAIN, 
			ALmixer_Channel_List[channel].fade_end_volume
		);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "39Testing error: %s\n",
			aluGetErrorString(error));				
	}

		return retval;
	}
	
	/* Enable fading effects via the flag */
	ALmixer_Channel_List[channel].fade_enabled = 1;
	/* Set fade start time */
	ALmixer_Channel_List[channel].fade_start_time
		= ALmixer_Channel_List[channel].start_time;
	fprintf(stderr, "Current time =%d\n", current_time);
	/* Set the fade expire ticks */
	ALmixer_Channel_List[channel].fade_expire_ticks = fade_ticks;

	/* Set 1/(endtime-starttime) or 1/deltaT */
	ALmixer_Channel_List[channel].fade_inv_time = 1.0f / fade_ticks;

	return retval;
	
}

		
static ALuint Internal_FadeInSourceTimed(ALuint source, ALmixer_Data* data, Sint32 loops, Uint32 fade_ticks, Sint32 expire_ticks)
{
	Sint32 channel;
	Sint32 retval;
	if(0 == source)
	{
		retval = Internal_FadeInChannelTimed(-1, data, loops, fade_ticks, expire_ticks);
		if(-1 == retval)
		{
			return 0;
		}
		else
		{
			return Internal_GetSource(retval);
		}
	}
	
	channel = Internal_GetChannel(source);
	if(-1 == channel)
	{
		ALmixer_SetError("Cannot FadeIn source: %s", ALmixer_GetError());
		return 0;
	}
	retval = Internal_FadeInChannelTimed(channel, data, loops, fade_ticks, expire_ticks);
	if(-1 == retval)
	{
		return 0;
	}
	else
	{
		return source;
	}
	/* make compiler happy */
	return 0;
}




/* Will fade out currently playing channels.
 * It starts at the current volume level and goes down */
static Sint32 Internal_FadeOutChannel(Sint32 channel, Uint32 ticks)
{
	ALfloat value;
	ALenum error;
	Uint32 current_time = SDL_GetTicks();
	Uint32 counter = 0;
	
	/* We can't accept 0 as a value because of div-by-zero.
	 * If zero, just call Halt at normal
	 * volume
	 */
	if(0 == ticks)
	{
		return Internal_HaltChannel(channel);
	}
	
	
	if(channel >= Number_of_Channels_global)
	{
			ALmixer_SetError("Requested channel (%d) exceeds maximum channel (%d) because only %d channels are allocated", channel, Number_of_Channels_global-1, Number_of_Channels_global);
		return -1;
	}

	if(channel >= 0)
	{
		if(ALmixer_Channel_List[channel].channel_in_use)
		{
			/* Get the current volume */
			alGetSourcef(ALmixer_Channel_List[channel].alsource,
				AL_MAX_GAIN, &value);
			ALmixer_Channel_List[channel].fade_start_volume = value;
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "40Testing error: %s\n",
			aluGetErrorString(error));				
	}
		
			/* Get the Min volume */
			alGetSourcef(ALmixer_Channel_List[channel].alsource,
				AL_MIN_GAIN, &value);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "41Testing error: %s\n",
			aluGetErrorString(error));				
	}
			ALmixer_Channel_List[channel].fade_end_volume = value;
			fprintf(stderr, "MIN gain: %f\n", value);
	
			/* Set expire start time */
			ALmixer_Channel_List[channel].start_time = current_time;
			/* Set the expire ticks */
			ALmixer_Channel_List[channel].expire_ticks = ticks;
			/* Set fade start time */
			ALmixer_Channel_List[channel].fade_start_time = current_time;
			/* Set the fade expire ticks */
			ALmixer_Channel_List[channel].fade_expire_ticks = ticks;
			/* Enable fading effects via the flag */
			ALmixer_Channel_List[channel].fade_enabled = 1;

			/* Set 1/(endtime-starttime) or 1/deltaT */
			ALmixer_Channel_List[channel].fade_inv_time = 1.0f / ticks;

			counter++;
		}
	}
	/* Else need to fade out all channels */
	else
	{
		Sint32 i;
		for(i=0; i<Number_of_Channels_global; i++)
		{
			if(ALmixer_Channel_List[i].channel_in_use)
			{
				/* Get the current volume */
				alGetSourcef(ALmixer_Channel_List[i].alsource,
					AL_MAX_GAIN, &value);
				ALmixer_Channel_List[i].fade_start_volume = value;
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "42Testing error: %s\n",
			aluGetErrorString(error));				
	}
			
				/* Get the Min volume */
				alGetSourcef(ALmixer_Channel_List[i].alsource,
					AL_MIN_GAIN, &value);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "43Testing error: %s\n",
			aluGetErrorString(error));				
	}
				ALmixer_Channel_List[i].fade_end_volume = value;
				fprintf(stderr, "MIN gain: %f\n", value);
	
				/* Set expire start time */
				ALmixer_Channel_List[i].start_time = current_time;
				/* Set the expire ticks */
				ALmixer_Channel_List[i].expire_ticks = ticks;
				/* Set fade start time */
				ALmixer_Channel_List[i].fade_start_time = current_time;
				/* Set the fade expire ticks */
				ALmixer_Channel_List[i].fade_expire_ticks = ticks;
				/* Enable fading effects via the flag */
				ALmixer_Channel_List[i].fade_enabled = 1;

				/* Set 1/(endtime-starttime) or 1/deltaT */
				ALmixer_Channel_List[i].fade_inv_time = 1.0f / ticks;

				counter++;
			}
		} /* End for loop */
	} 
	return counter;
}


static Sint32 Internal_FadeOutSource(ALuint source, Uint32 ticks)
{
	Sint32 channel;
	if(0 == source)
	{
		return Internal_FadeOutChannel(-1, ticks);
	}
	
	channel = Internal_GetChannel(source);
	if(-1 == channel)
	{
		ALmixer_SetError("Cannot FadeOut source: %s", ALmixer_GetError());
		return -1;
	}
	return Internal_FadeOutChannel(channel, ticks);
}


/* Will fade currently playing channels.
 * It starts at the current volume level and go to target 
 * Only affects channels that are playing
 */
static Sint32 Internal_FadeChannel(Sint32 channel, Uint32 ticks, ALfloat volume)
{
	ALfloat value;
	ALenum error;
	Uint32 current_time = SDL_GetTicks();
	Uint32 counter = 0;
	
	if(channel >= Number_of_Channels_global)
	{
			ALmixer_SetError("Requested channel (%d) exceeds maximum channel (%d) because only %d channels are allocated", channel, Number_of_Channels_global-1, Number_of_Channels_global);
		return -1;
	}

	if(channel >= 0)
	{
		if(volume < ALmixer_Channel_List[channel].min_volume)
		{
			volume = ALmixer_Channel_List[channel].min_volume;
		}
		else if(volume > ALmixer_Channel_List[channel].max_volume)
		{
			volume = ALmixer_Channel_List[channel].max_volume;
		}
		
		if(ALmixer_Channel_List[channel].channel_in_use)
		{
			if(ticks > 0)
			{
				/* Get the current volume */
				alGetSourcef(ALmixer_Channel_List[channel].alsource,
					AL_MAX_GAIN, &value);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "44Testing error: %s\n",
			aluGetErrorString(error));				
	}
				ALmixer_Channel_List[channel].fade_start_volume = value;
			
				/* Set the target volume */
				ALmixer_Channel_List[channel].fade_end_volume = volume;
		
				/* Set fade start time */
				ALmixer_Channel_List[channel].fade_start_time = current_time;
				/* Set the fade expire ticks */
				ALmixer_Channel_List[channel].fade_expire_ticks = ticks;
				/* Enable fading effects via the flag */
				ALmixer_Channel_List[channel].fade_enabled = 1;

				/* Set 1/(endtime-starttime) or 1/deltaT */
				ALmixer_Channel_List[channel].fade_inv_time = 1.0f / ticks;
			}
			else
			{
				alSourcef(ALmixer_Channel_List[channel].alsource,
					AL_MAX_GAIN, volume);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "45Testing error: %s\n",
			aluGetErrorString(error));				
	}
			}
			counter++;
		}
	}
	/* Else need to fade out all channels */
	else
	{
		Sint32 i;
		for(i=0; i<Number_of_Channels_global; i++)
		{
			if(volume < ALmixer_Channel_List[i].min_volume)
			{
				volume = ALmixer_Channel_List[i].min_volume;
			}
			else if(volume > ALmixer_Channel_List[i].max_volume)
			{
				volume = ALmixer_Channel_List[i].max_volume;
			}
		
			if(ALmixer_Channel_List[i].channel_in_use)
			{
				if(ticks > 0)
				{
					/* Get the current volume */
					alGetSourcef(ALmixer_Channel_List[i].alsource,
						AL_MAX_GAIN, &value);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "46Testing error: %s\n",
			aluGetErrorString(error));				
	}
					ALmixer_Channel_List[i].fade_start_volume = value;
				
					/* Set target volume */
					ALmixer_Channel_List[i].fade_end_volume = volume;
		
					/* Set fade start time */
					ALmixer_Channel_List[i].fade_start_time = current_time;
					/* Set the fade expire ticks */
					ALmixer_Channel_List[i].fade_expire_ticks = ticks;
					/* Enable fading effects via the flag */
					ALmixer_Channel_List[i].fade_enabled = 1;

					/* Set 1/(endtime-starttime) or 1/deltaT */
					ALmixer_Channel_List[i].fade_inv_time = 1.0f / ticks;
				}
				else
				{
					alSourcef(ALmixer_Channel_List[i].alsource,
						AL_MAX_GAIN, volume);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "47Testing error: %s\n",
			aluGetErrorString(error));				
	}
				}
				counter++;
			}
		} /* End for loop */
	} 
	return counter;
}

static Sint32 Internal_FadeSource(ALuint source, Uint32 ticks, ALfloat volume)
{
	Sint32 channel;
	if(0 == source)
	{
		return Internal_FadeChannel(-1, ticks, volume);
	}
	
	channel = Internal_GetChannel(source);
	if(-1 == channel)
	{
		ALmixer_SetError("Cannot Fade source: %s", ALmixer_GetError());
		return -1;
	}
	return Internal_FadeChannel(channel, ticks, volume);
}




/* Set a volume regardless if it's in use or not.
 */
static Sint32 Internal_SetMaxVolumeChannel(Sint32 channel, ALfloat volume)
{
	ALenum error;
	Sint32 retval = 0;
	
	if(channel >= Number_of_Channels_global)
	{
			ALmixer_SetError("Requested channel (%d) exceeds maximum channel (%d) because only %d channels are allocated", channel, Number_of_Channels_global-1, Number_of_Channels_global);
		return -1;
	}

	if(channel >= 0)
	{
		if(volume < 0.0f)
		{
			volume = 0.0f;
		}
		else if(volume > 1.0f)
		{
			volume = 1.0f;
		}
		ALmixer_Channel_List[channel].max_volume = volume;
		alSourcef(ALmixer_Channel_List[channel].alsource,
			AL_MAX_GAIN, volume);
		if((error = alGetError()) != AL_NO_ERROR)
		{
			ALmixer_SetError("%s",
				aluGetErrorString(error) );
			retval = -1;
		}
		if(ALmixer_Channel_List[channel].max_volume < ALmixer_Channel_List[channel].min_volume)
		{
			ALmixer_Channel_List[channel].min_volume = volume;
			alSourcef(ALmixer_Channel_List[channel].alsource,
				AL_MIN_GAIN, volume);
			if((error = alGetError()) != AL_NO_ERROR)
			{
				ALmixer_SetError("%s",
					aluGetErrorString(error) );
				retval = -1;
			}
		}
	}
	else
	{
		Sint32 i;
		for(i=0; i<Number_of_Channels_global; i++)
		{
			if(volume < 0.0f)
			{
				volume = 0.0f;
			}
			else if(volume > 1.0f)
			{
				volume = 1.0f;
			}
			ALmixer_Channel_List[i].max_volume = volume;
			alSourcef(ALmixer_Channel_List[i].alsource,
				AL_MAX_GAIN, volume);
			if((error = alGetError()) != AL_NO_ERROR)
			{
				ALmixer_SetError("%s",
					aluGetErrorString(error) );
				retval = -1;
			}
			if(ALmixer_Channel_List[i].max_volume < ALmixer_Channel_List[i].min_volume)
			{
				ALmixer_Channel_List[i].min_volume = volume;
				alSourcef(ALmixer_Channel_List[i].alsource,
					AL_MIN_GAIN, volume);
				if((error = alGetError()) != AL_NO_ERROR)
				{
					ALmixer_SetError("%s",
						aluGetErrorString(error) );
					retval = -1;
				}
			}
		}
	}
	return retval;
}

static Sint32 Internal_SetMaxVolumeSource(ALuint source, ALfloat volume)
{
	Sint32 channel;
	if(0 == source)
	{
		return Internal_SetMaxVolumeChannel(-1, volume);
	}
	
	channel = Internal_GetChannel(source);
	if(-1 == channel)
	{
		ALmixer_SetError("Cannot SetMaxVolume: %s", ALmixer_GetError());
		return -1;
	}
	return Internal_SetMaxVolumeChannel(channel, volume);
}

static ALfloat Internal_GetMaxVolumeChannel(Sint32 channel)
{
	/*
	ALfloat value;
	ALenum error;
	*/
	ALfloat running_total = 0.0f;
	ALfloat retval = 0.0f;
	
	if(channel >= Number_of_Channels_global)
	{
			ALmixer_SetError("Requested channel (%d) exceeds maximum channel (%d) because only %d channels are allocated", channel, Number_of_Channels_global-1, Number_of_Channels_global);
		return -1.0f;
	}

	if(channel >= 0)
	{
		/*
		alGetSourcef(ALmixer_Channel_List[channel].alsource,
			AL_GAIN, &value);
		if((error = alGetError()) != AL_NO_ERROR)
		{
			ALmixer_SetError("%s",
				aluGetErrorString(error) );
			retval = -1.0f;
		}
		else
		{
			retval = value;
		}
		*/
		retval = ALmixer_Channel_List[channel].max_volume;
	
	}
	else
	{
		Sint32 i;
		for(i=0; i<Number_of_Channels_global; i++)
		{
			/*
			alGetSourcef(ALmixer_Channel_List[i].alsource,
				AL_GAIN, &value);
			if((error = alGetError()) != AL_NO_ERROR)
			{
				ALmixer_SetError("%s",
					aluGetErrorString(error) );
				retval = -1;
			}
			else
			{
				running_total += value;
			}
			*/
			running_total += ALmixer_Channel_List[i].max_volume;
		}
		if(0 == Number_of_Channels_global)
		{
			ALmixer_SetError("No channels are allocated");
			retval = -1.0f;
		}
		else
		{
			retval = running_total / Number_of_Channels_global;
		}
	}
	return retval;
}

static ALfloat Internal_GetMaxVolumeSource(ALuint source)
{
	Sint32 channel;
	if(0 == source)
	{
		return Internal_GetMaxVolumeChannel(-1);
	}
	
	channel = Internal_GetChannel(source);
	if(-1 == channel)
	{
		ALmixer_SetError("Cannot GetVolume: %s", ALmixer_GetError());
		return -1.0f;
	}

	return Internal_GetMaxVolumeChannel(channel);
}


/* Set a volume regardless if it's in use or not.
 */
static Sint32 Internal_SetMinVolumeChannel(Sint32 channel, ALfloat volume)
{
	ALenum error;
	Sint32 retval = 0;
	
	if(channel >= Number_of_Channels_global)
	{
			ALmixer_SetError("Requested channel (%d) exceeds maximum channel (%d) because only %d channels are allocated", channel, Number_of_Channels_global-1, Number_of_Channels_global);
		return -1;
	}

	if(channel >= 0)
	{
		if(volume < 0.0f)
		{
			volume = 0.0f;
		}
		else if(volume > 1.0f)
		{
			volume = 1.0f;
		}
		ALmixer_Channel_List[channel].min_volume = volume;
		alSourcef(ALmixer_Channel_List[channel].alsource,
			AL_MIN_GAIN, volume);
		if((error = alGetError()) != AL_NO_ERROR)
		{
			ALmixer_SetError("%s",
				aluGetErrorString(error) );
			retval = -1;
		}
		if(ALmixer_Channel_List[channel].max_volume < ALmixer_Channel_List[channel].min_volume)
		{
			ALmixer_Channel_List[channel].max_volume = volume;
			alSourcef(ALmixer_Channel_List[channel].alsource,
				AL_MAX_GAIN, volume);
			if((error = alGetError()) != AL_NO_ERROR)
			{
				ALmixer_SetError("%s",
					aluGetErrorString(error) );
				retval = -1;
			}
		}
	}
	else
	{
		Sint32 i;
		for(i=0; i<Number_of_Channels_global; i++)
		{
			if(volume < 0.0f)
			{
				volume = 0.0f;
			}
			else if(volume > 1.0f)
			{
				volume = 1.0f;
			}
			ALmixer_Channel_List[i].min_volume = volume;
			alSourcef(ALmixer_Channel_List[i].alsource,
				AL_MIN_GAIN, volume);
			if((error = alGetError()) != AL_NO_ERROR)
			{
				ALmixer_SetError("%s",
					aluGetErrorString(error) );
				retval = -1;
			}
			if(ALmixer_Channel_List[i].max_volume < ALmixer_Channel_List[i].min_volume)
			{
				ALmixer_Channel_List[i].max_volume = volume;
				alSourcef(ALmixer_Channel_List[i].alsource,
					AL_MAX_GAIN, volume);
				if((error = alGetError()) != AL_NO_ERROR)
				{
					ALmixer_SetError("%s",
						aluGetErrorString(error) );
					retval = -1;
				}
			}
		}
	}
	return retval;
}

static Sint32 Internal_SetMinVolumeSource(ALuint source, ALfloat volume)
{
	Sint32 channel;
	if(0 == source)
	{
		return Internal_SetMinVolumeChannel(-1, volume);
	}
	
	channel = Internal_GetChannel(source);
	if(-1 == channel)
	{
		ALmixer_SetError("Cannot SetMaxVolume: %s", ALmixer_GetError());
		return -1;
	}
	return Internal_SetMinVolumeChannel(channel, volume);
}

static ALfloat Internal_GetMinVolumeChannel(Sint32 channel)
{
	/*
	ALfloat value;
	ALenum error;
	*/
	ALfloat running_total = 0.0f;
	ALfloat retval = 0.0f;
	
	if(channel >= Number_of_Channels_global)
	{
			ALmixer_SetError("Requested channel (%d) exceeds maximum channel (%d) because only %d channels are allocated", channel, Number_of_Channels_global-1, Number_of_Channels_global);
		return -1.0f;
	}

	if(channel >= 0)
	{
		/*
		alGetSourcef(ALmixer_Channel_List[channel].alsource,
			AL_GAIN, &value);
		if((error = alGetError()) != AL_NO_ERROR)
		{
			ALmixer_SetError("%s",
				aluGetErrorString(error) );
			retval = -1.0f;
		}
		else
		{
			retval = value;
		}
		*/
		retval = ALmixer_Channel_List[channel].min_volume;
	
	}
	else
	{
		Sint32 i;
		for(i=0; i<Number_of_Channels_global; i++)
		{
			/*
			alGetSourcef(ALmixer_Channel_List[i].alsource,
				AL_GAIN, &value);
			if((error = alGetError()) != AL_NO_ERROR)
			{
				ALmixer_SetError("%s",
					aluGetErrorString(error) );
				retval = -1;
			}
			else
			{
				running_total += value;
			}
			*/
			running_total += ALmixer_Channel_List[i].min_volume;
		}
		if(0 == Number_of_Channels_global)
		{
			ALmixer_SetError("No channels are allocated");
			retval = -1.0f;
		}
		else
		{
			retval = running_total / Number_of_Channels_global;
		}
	}
	return retval;
}

static ALfloat Internal_GetMinVolumeSource(ALuint source)
{
	Sint32 channel;
	if(0 == source)
	{
		return Internal_GetMinVolumeChannel(-1);
	}
	
	channel = Internal_GetChannel(source);
	if(-1 == channel)
	{
		ALmixer_SetError("Cannot GetVolume: %s", ALmixer_GetError());
		return -1.0f;
	}

	return Internal_GetMinVolumeChannel(channel);
}


/* Changes the listener volume */
static Sint32 Internal_SetMasterVolume(ALfloat volume)
{
	ALenum error;
	alListenerf(AL_GAIN, volume);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		ALmixer_SetError("%s",
			aluGetErrorString(error) );
		return -1;
	}
	return 0;	
}

static ALfloat Internal_GetMasterVolume()
{
	ALenum error;
	ALfloat volume;
	alGetListenerf(AL_GAIN, &volume);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		ALmixer_SetError("%s",
			aluGetErrorString(error) );
		return -1.0f;
	}
	return volume;	
}




/* Will fade out currently playing channels.
 * It starts at the current volume level and goes down */
static Sint32 Internal_ExpireChannel(Sint32 channel, Sint32 ticks)
{
	Uint32 current_time = SDL_GetTicks();
	Uint32 counter = 0;
	
	/* We can't accept 0 as a value because of div-by-zero.
	 * If zero, just call Halt at normal
	 * volume
	 */
	if(0 == ticks)
	{
		return Internal_HaltChannel(channel);
	}
	if(ticks < -1)
	{
		ticks = -1;
	}
	
	
	if(channel >= Number_of_Channels_global)
	{
			ALmixer_SetError("Requested channel (%d) exceeds maximum channel (%d) because only %d channels are allocated", channel, Number_of_Channels_global-1, Number_of_Channels_global);
		return -1;
	}

	if(channel >= 0)
	{
		if(ALmixer_Channel_List[channel].channel_in_use)
		{
			/* Set expire start time */
			ALmixer_Channel_List[channel].start_time = current_time;
			/* Set the expire ticks */
			ALmixer_Channel_List[channel].expire_ticks = ticks;

			counter++;
		}
	}
	/* Else need to fade out all channels */
	else
	{
		Sint32 i;
		for(i=0; i<Number_of_Channels_global; i++)
		{
			if(ALmixer_Channel_List[i].channel_in_use)
			{
				/* Set expire start time */
				ALmixer_Channel_List[i].start_time = current_time;
				/* Set the expire ticks */
				ALmixer_Channel_List[i].expire_ticks = ticks;

				counter++;
			}
		} /* End for loop */
	} 
	return counter;
}


static Sint32 Internal_ExpireSource(ALuint source, Sint32 ticks)
{
	Sint32 channel;
	if(0 == source)
	{
		return Internal_ExpireChannel(-1, ticks);
	}
	
	channel = Internal_GetChannel(source);
	if(-1 == channel)
	{
		ALmixer_SetError("Cannot Expire source: %s", ALmixer_GetError());
		return -1;
	}
	return Internal_ExpireChannel(channel, ticks);
}


static Sint32 Internal_QueryChannel(Sint32 channel)
{
	Sint32 i;
	Sint32 counter = 0;
	if(channel >= Number_of_Channels_global)
	{
		ALmixer_SetError("Invalid channel: %d", channel);
		return -1;
	}

	if(channel >= 0)
	{
		return ALmixer_Channel_List[channel].channel_in_use;	
	}

	/* Else, return the number of channels in use */
	for(i=0; i<Number_of_Channels_global; i++)
	{
		if(ALmixer_Channel_List[i].channel_in_use)
		{
			counter++;
		}
	}
	return counter;
}


static Sint32 Internal_QuerySource(ALuint source)
{
	Sint32 channel;
	if(0 == source)
	{
		return Internal_QueryChannel(-1);
	}
	
	channel = Internal_GetChannel(source);
	if(-1 == channel)
	{
		ALmixer_SetError("Cannot query source: %s", ALmixer_GetError());
		return -1;
	}
	
	return Internal_QueryChannel(channel);
}


static Sint32 Internal_CountUnreservedUsedChannels()
{
	Sint32 i;
	Sint32 counter = 0;


	/* Else, return the number of channels in use */
	for(i=Number_of_Reserve_Channels_global; i<Number_of_Channels_global; i++)
	{
		if(ALmixer_Channel_List[i].channel_in_use)
		{
			counter++;
		}
	}
	return counter;
}

static Sint32 Internal_CountUnreservedFreeChannels()
{
	Sint32 i;
	Sint32 counter = 0;


	/* Else, return the number of channels in use */
	for(i=Number_of_Reserve_Channels_global; i<Number_of_Channels_global; i++)
	{
		if( ! ALmixer_Channel_List[i].channel_in_use)
		{
			counter++;
		}
	}
	return counter;
}

static Sint32 Internal_CountAllUsedChannels()
{
	Sint32 i;
	Sint32 counter = 0;


	/* Else, return the number of channels in use */
	for(i=0; i<Number_of_Channels_global; i++)
	{
		if(ALmixer_Channel_List[i].channel_in_use)
		{
			counter++;
		}
	}
	return counter;
}

static Sint32 Internal_CountAllFreeChannels()
{
	Sint32 i;
	Sint32 counter = 0;


	/* Else, return the number of channels in use */
	for(i=0; i<Number_of_Channels_global; i++)
	{
		if( ! ALmixer_Channel_List[i].channel_in_use)
		{
			counter++;
		}
	}
	return counter;
}


static Sint32 Internal_PlayingChannel(Sint32 channel)
{
	Sint32 i;
	Sint32 counter = 0;
	ALint state;
	
	if(channel >= Number_of_Channels_global)
	{
		ALmixer_SetError("Invalid channel: %d", channel);
		return -1;
	}

	if(channel >= 0)
	{
		if(ALmixer_Channel_List[channel].channel_in_use)	
		{
			alGetSourcei(
				ALmixer_Channel_List[channel].alsource,
				AL_SOURCE_STATE, &state
			);
			if(AL_PLAYING == state)
			{
				return 1;
			}
		}
		return 0;
	}

	/* Else, return the number of channels in use */
	for(i=0; i<Number_of_Channels_global; i++)
	{
		if(ALmixer_Channel_List[i].channel_in_use)	
		{
			alGetSourcei(
				ALmixer_Channel_List[i].alsource,
				AL_SOURCE_STATE, &state
			);
			if(AL_PLAYING == state)
			{
				counter++;
			}
		}
	}
	return counter;
}


static Sint32 Internal_PlayingSource(ALuint source)
{
	Sint32 channel;
	if(0 == source)
	{
		return Internal_PlayingChannel(-1);
	}
	
	channel = Internal_GetChannel(source);
	if(-1 == channel)
	{
		ALmixer_SetError("Cannot query source: %s", ALmixer_GetError());
		return -1;
	}
	
	return Internal_PlayingChannel(channel);
}


static Sint32 Internal_PausedChannel(Sint32 channel)
{
	Sint32 i;
	Sint32 counter = 0;
	ALint state;
	
	if(channel >= Number_of_Channels_global)
	{
		ALmixer_SetError("Invalid channel: %d", channel);
		return -1;
	}

	if(channel >= 0)
	{
		if(ALmixer_Channel_List[channel].channel_in_use)	
		{
			alGetSourcei(
				ALmixer_Channel_List[channel].alsource,
				AL_SOURCE_STATE, &state
			);
			if(AL_PAUSED == state)
			{
				return 1;
			}
		}
		return 0;
	}

	/* Else, return the number of channels in use */
	for(i=0; i<Number_of_Channels_global; i++)
	{
		if(ALmixer_Channel_List[i].channel_in_use)	
		{
			alGetSourcei(
				ALmixer_Channel_List[i].alsource,
				AL_SOURCE_STATE, &state
			);
			if(AL_PAUSED == state)
			{
				counter++;
			}
		}
	}
	return counter;
}


static Sint32 Internal_PausedSource(ALuint source)
{
	Sint32 channel;
	if(0 == source)
	{
		return Internal_PausedChannel(-1);
	}
	
	channel = Internal_GetChannel(source);
	if(-1 == channel)
	{
		ALmixer_SetError("Cannot query source: %s", ALmixer_GetError());
		return -1;
	}
	
	return Internal_PausedChannel(channel);
}





				
/* Private function for Updating ALmixer.
 * This is a very big and ugly function.
 * It should return the number of buffers that were 
 * queued during the call. The value might be
 * used to guage how long you might wait to
 * call the next update loop in case you are worried
 * about preserving CPU cycles. The idea is that
 * when a buffer is queued, there was probably some
 * CPU intensive looping which took awhile.
 * It's mainly provided as a convenience.
 * Timing the call with SDL_GetTicks() would produce
 * more accurate information.
 * Returns a negative value if there was an error,
 * the value being the number of errors.
 */
static Sint32 Update_ALmixer(void* data)
{
	Sint32 retval = 0;
	Sint32 error_flag = 0;
	ALenum error;
	ALint state;
	Sint32 i=0;

	SDL_LockMutex(simple_lock);
	if(0 == ALmixer_Initialized)
	{
		SDL_UnlockMutex(simple_lock);
		return 0;
	}
	
	/* Check the quick flag to see if anything needs updating */
	/* If anything is playing, then we have to do work */
	if( 0 == Is_Playing_global)
	{
		SDL_UnlockMutex(simple_lock);
		return 0;
	}
	/* Clear error */
	if((error = alGetError()) != AL_NO_ERROR)
				{
		fprintf(stderr, "08Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n",
				aluGetErrorString(error));				
				}
	alGetError();

	for(i=0; i<Number_of_Channels_global; i++)
	{
		if( ALmixer_Channel_List[i].channel_in_use )
		{

			/* For simplicity, before we do anything else,
			 * we can check the timeout and fading values
			 * and do the appropriate things
			 */
			Uint32 current_time = SDL_GetTicks();

			/* Check to see if we need to halt due to Timed play */
			if(ALmixer_Channel_List[i].expire_ticks != -1) 
			{
				Uint32 target_time = (Uint32)ALmixer_Channel_List[i].expire_ticks 
					 + ALmixer_Channel_List[i].start_time;
				alGetSourcei(ALmixer_Channel_List[i].alsource,
					AL_SOURCE_STATE, &state);
				if((error = alGetError()) != AL_NO_ERROR)
				{
					fprintf(stderr, "06Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n",
							aluGetErrorString(error));				
				}
				
				/* Check the time, and also make sure that it is not
				 * paused (if paused, we don't want to make the
				 * evaluation because when resumed, we will adjust
				 * the times to compensate for the pause).
				 */
				if( (current_time >= target_time) 
					&& (state != AL_PAUSED) )
				{
					/* Stop the playback */
					ALmixer_HaltChannel(i);
					if((error = alGetError()) != AL_NO_ERROR)
					{
						fprintf(stderr, "07Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n",
								aluGetErrorString(error));				
					}
					
					/* Everything should be done so go on to the next loop */
					continue;
				}
			} /* End if time expired check */

			/* Check to see if we need to adjust the volume for fading */
			if( ALmixer_Channel_List[i].fade_enabled )
			{
				Uint32 target_time = ALmixer_Channel_List[i].fade_expire_ticks 
					 + ALmixer_Channel_List[i].fade_start_time;
				alGetSourcei(ALmixer_Channel_List[i].alsource,
					AL_SOURCE_STATE, &state);
				if((error = alGetError()) != AL_NO_ERROR)
				{
					fprintf(stderr, "05Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n",
							aluGetErrorString(error));				
				}
				
				/* Check the time, and also make sure that it is not
				 * paused (if paused, we don't want to make the
				 * evaluation because when resumed, we will adjust
				 * the times to compensate for the pause).
				 */
				if(state != AL_PAUSED)
				{
					ALfloat t;
					Uint32 delta_time;
					ALfloat current_volume;
					if(current_time >= target_time)
					{
						/* Need to constrain value to the end time
						 * (can't go pass the value for calculations)
						 */
						current_time = target_time;
						/* We can disable the fade flag now */
						ALmixer_Channel_List[i].fade_enabled = 0;
					}
					/* Use the linear interpolation formula:
					 * X = (1-t)x0 + tx1
					 * where x0 would be the start value
					 * and x1 is the final value
					 * and t is delta_time*inv_time (adjusts 0 <= time <= 1)
					 * delta_time = current_time-start_time
					 * inv_time = 1/ (end_time-start_time)
					 * so t = current_time-start_time / (end_time-start_time)
					 * 
					 */
					delta_time = current_time - ALmixer_Channel_List[i].fade_start_time;
					t = (ALfloat) delta_time * ALmixer_Channel_List[i].fade_inv_time;

					current_volume = (1.0f-t) * ALmixer_Channel_List[i].fade_start_volume 
						+ t * ALmixer_Channel_List[i].fade_end_volume;

					/* Set the volume */
					alSourcef(ALmixer_Channel_List[i].alsource,
						AL_MAX_GAIN, current_volume);
					if((error = alGetError()) != AL_NO_ERROR)
					{
						fprintf(stderr, "04Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n",
								aluGetErrorString(error));				
					}
					
	/*
	fprintf(stderr, "Current time =%d\n", current_time);
						fprintf(stderr, "Current vol=%f on channel %d\n", current_volume, i);
	*/
				} /* End if not PAUSED */
			} /* End if fade_enabled */
					 

			/* Okay, now that the time expired and fading stuff
			 * is done, do the rest of the hard stuff
			 */
			

			/* For predecoded, check to see if done */
			if( ALmixer_Channel_List[i].almixer_data->decoded_all ) 
			{

#if 0
		/********* Remove this **********/
				ALint buffers_processed;
				ALint buffers_still_queued;
		fprintf(stderr, "For Predecoded\n");
				
	alGetSourcei(
		ALmixer_Channel_List[i].alsource,
		AL_SOURCE_STATE, &state
				);
	switch(state) {
                case AL_PLAYING:
				fprintf(stderr, "Channel '%d' is PLAYING\n", i);
				break;
                case AL_PAUSED:
				fprintf(stderr, "Channel '%d' is PAUSED\n",i);
				break;
                case AL_STOPPED:
				fprintf(stderr, "Channel '%d' is STOPPED\n",i);
				break;
				case AL_INITIAL:
				fprintf(stderr, "Channel '%d' is INITIAL\n",i);
				break;
                default:
				fprintf(stderr, "Channel '%d' is UNKNOWN\n",i);
                  break;
       		}
					
				alGetSourcei(
					ALmixer_Channel_List[i].alsource,
					AL_BUFFERS_PROCESSED, &buffers_processed
				);
				fprintf(stderr, "Buffers processed = %d\n", buffers_processed);

				alGetSourcei(
					ALmixer_Channel_List[i].alsource,
					AL_BUFFERS_QUEUED, &buffers_still_queued
				);

				/******** END REMOVE *******/
#endif	
				/* FIXME: Ugh! Somewhere an alError is being thrown ("Invalid Enum Value"), but I can't
				 * find it. It only seems to be thrown for OS X. I placed error messages after every al*
				 * command I could find in the above loops, but the error doesn't seem to show 
				 * up until around here. I mistook it for a get queued buffers
				 * error in OS X. I don't think there's an error down there. 
				 * For now, I'm clearing the error here.
				 */
					
				if((error = alGetError()) != AL_NO_ERROR)
				{
					fprintf(stderr, "03Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n",
							aluGetErrorString(error));				
				}
				
					
				alGetSourcei(
					ALmixer_Channel_List[i].alsource,
					AL_SOURCE_STATE, &state
				);
				if((error = alGetError()) != AL_NO_ERROR)
				{
					fprintf(stderr, "02Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n",
							aluGetErrorString(error));				
				}
				
				
				if(AL_STOPPED == state)
				{
					/* Playback has ended. 
					 * Loop if necessary, or launch callback
					 * and clear channel (or clear channel and
					 * then launch callback?)
					 */


					/* Need to check for loops */
					if(ALmixer_Channel_List[i].loops != 0)
					{
						/* Corner Case: If the buffer has
						 * been modified using Seek,
						 * the loop will start at the seek
						 * position.
						 */
						if(ALmixer_Channel_List[i].loops != -1)
						{
							ALmixer_Channel_List[i].loops--;
						}
						alSourcePlay(ALmixer_Channel_List[i].alsource);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "50Testing error: %s\n",
			aluGetErrorString(error));				
	}
						continue;
					}
					/* No loops. End play. */
					else
					{
						/* Problem: It seems that when mixing
						 * streamed and predecoded sources,
						 * the previous instance lingers, 
						 * so we need to force remove
						 * the data from the source.
			 			 * The sharing problem
			 			 * occurs when a previous predecoded buffer is played on
			 			 * a source, and then a streamed source is played later
			 			 * on that same source. OpenAL isn't consistently
			 			 * removing the previous buffer so both get played.
			 			 * (Different dists seem to have different quirks.
			 			 * The problem might lead to crashes in the worst case.)
						 */
						/* Additional problem: There is another 
						 * inconsistency among OpenAL distributions.
						 * Both Loki and Creative Windows seem to keep
						 * the buffer queued which requires removing.
						 * But the Creative Macintosh version does
						 * not have any buffer queued after play
						 * and it returns the error: Invalid Enum Value
						 * if I try to unqueue it.
						 * So I'm going to put in a check to see if I
						 * can detect any buffers queued first
				 		 * and then unqueue them if I can see them.
						 * Additional note: The new CoreAudio based 
						 * implementation leaves it's buffer queued
						 * like Loki and Creative Windows. But 
						 * considering all the problems I'm having
						 * with the different distributions, this
						 * check seems reasonable.
						 */
						ALint buffers_still_queued;
						if((error = alGetError()) != AL_NO_ERROR)
						{
							fprintf(stderr, "01Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n",
									aluGetErrorString(error));				
						}
						
						alGetSourcei(
							ALmixer_Channel_List[i].alsource,
							AL_BUFFERS_QUEUED, &buffers_still_queued
						);
						if((error = alGetError()) != AL_NO_ERROR)
						{
		fprintf(stderr, "Error with unqueue, for OS X this is expected: %s\n",
							aluGetErrorString(error));
							ALmixer_SetError("Failed detecting unqueued predecoded buffer (expected with OS X): %s",
								aluGetErrorString(error) );
							error_flag--;
						}
						if(buffers_still_queued > 0)
						{
							/*
							PrintQueueStatus(ALmixer_Channel_List[i].alsource);
							*/
							
							/* Rather than force unqueuing the buffer, let's see if
							 * setting the buffer to none works (the OpenAL 1.0 
							 * Reference Annotation suggests this should work).							 
							 */
							alSourcei(ALmixer_Channel_List[i].alsource,
										AL_BUFFER, AL_NONE); 
							/*
							PrintQueueStatus(ALmixer_Channel_List[i].alsource);
							*/	
							if((error = alGetError()) != AL_NO_ERROR)
							{
		fprintf(stderr, "Error with unqueue, after alSourceUnqueueBuffers, buffers_still_queued=%d, error is: %s", buffers_still_queued,
								aluGetErrorString(error));
								ALmixer_SetError("Predecoded Unqueue buffer failed: %s",
									aluGetErrorString(error) );
								error_flag--;
							}

						}
						
						Clean_Channel(i);
						/* Subtract counter */
						Is_Playing_global--;

						/* Launch callback */
						Invoke_Channel_Done_Callback(i);

						/* We're done for this loop.
						 * Go to next channel 
						 */
						continue;
					}
					continue;
				}
			} /* End if decoded_all */
			/* For streamed */
			else
			{
				ALint buffers_processed;
				ALint buffers_still_queued;
				ALint current_buffer_id;

				ALuint unqueued_buffer_id;
#if 0
		/********* Remove this **********/
		fprintf(stderr, "For Streamed\n");
				
	alGetSourcei(
		ALmixer_Channel_List[i].alsource,
		AL_SOURCE_STATE, &state
				);
	switch(state) {
                case AL_PLAYING:
				fprintf(stderr, "Channel '%d' is PLAYING\n", i);
				break;
                case AL_PAUSED:
				fprintf(stderr, "Channel '%d' is PAUSED\n",i);
				break;
                case AL_STOPPED:
				fprintf(stderr, "Channel '%d' is STOPPED\n",i);
				break;
				case AL_INITIAL:
				fprintf(stderr, "Channel '%d' is INITIAL\n",i);
				break;
                default:
				fprintf(stderr, "Channel '%d' is UNKNOWN\n",i);
                  break;
       		}
				/******** END REMOVE *******/
#endif	
				/* Get the number of buffers still queued */
				alGetSourcei(
					ALmixer_Channel_List[i].alsource,
					AL_BUFFERS_QUEUED, &buffers_still_queued
				);
				if((error = alGetError()) != AL_NO_ERROR)
				{
					fprintf(stderr, "51Testing error: %s\n",
						aluGetErrorString(error));				
				}
				/* Get the number of buffers processed
				 * so we know if we need to refill 
				 */
				alGetSourcei(
					ALmixer_Channel_List[i].alsource,
					AL_BUFFERS_PROCESSED, &buffers_processed
				);
				if((error = alGetError()) != AL_NO_ERROR)
				{
					fprintf(stderr, "52Testing error: %s\n",
						aluGetErrorString(error));				
				}

 /* WTF!!! The Nvidia distribution is failing on the alGetSourcei(source, AL_BUFFER, buf_id) call.
 * I need this call to figure out which buffer OpenAL is currently playing. 
 * It keeps returning an "Invalid Enum" error.
 * This is totally inane! It's a basic query.
 * By the spec, this functionality is not explicitly defined so Nvidia refuses to
 * fix this behavior, even though all other distributions work fine with this.
 * The only workaround for this is for
 * a significant rewrite of my code which requires me to
 * duplicate the OpenAL queued buffers state with my own
 * code and try to derive what the current playing buffer is by indirect observation of 
 * looking at buffers_processed. But of course this has a ton of downsides since my 
 * queries do not give me perfect timing of what OpenAL is actually doing and
 * the fact that some of the distributions seem to have buffer queuing problems
 * with their query results (CoreAudio).  This also means a ton of extra code
 * on my side. The lack of support of a 1 line call has required me to
 * implement yet another entire state machine. <sigh>
 */
#if 0 /* This code will not work until possibly OpenAL 1.1 because of Nvidia */
				/* Get the id to the current buffer playing */
				alGetSourcei(
					ALmixer_Channel_List[i].alsource,
					AL_BUFFER, &current_buffer_id
				);
				if((error = alGetError()) != AL_NO_ERROR)
				{
					fprintf(stderr, "53Testing error: %s\n",
						aluGetErrorString(error));				
				}

				/* Before the hard stuff, check to see if the 
				 * current queued AL buffer has changed.
				 * If it has, we should launch a data callback if
				 * necessary
				 */
				if( ((ALuint)current_buffer_id) != 
					ALmixer_Channel_List[i].almixer_data->current_buffer)
				{
					ALmixer_Channel_List[i].almixer_data->current_buffer 
						= (ALuint)current_buffer_id;

					Invoke_Streamed_Channel_Data_Callback(i, ALmixer_Channel_List[i].almixer_data, current_buffer_id);
				}
#else
				/* Only do this if "access_data" was requested (i.e. the circular_buffer!=NULL) 
				 * And if one of the two are true:
				 * Either buffers_processed > 0 (because the current_buffer might have changed)
				 * or if the current_buffer==0 (because we are in an initial state or recovering from 
				 * a buffer underrun)
				 */
				if((ALmixer_Channel_List[i].almixer_data->circular_buffer_queue != NULL) 
				   && (
							(buffers_processed > 0) || (0 == ALmixer_Channel_List[i].almixer_data->current_buffer)
					)
				   )
				{
					Uint32 k;
					Uint32 queue_ret_flag;
					Uint8 is_out_of_sync = 0;
					Uint32 my_queue_size = CircularQueueUnsignedInt_Size(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue);
/*
					fprintf(stderr, "Queue in processed check, before pop, buffers_processed=%d\n", buffers_processed);
					CircularQueueUnsignedInt_Print(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue);
*/
					/* We can't make any determinations solely based on the number of buffers_processed
					 * because currently, we only unqueue 1 buffer per loop. That means if 2 or more
					 * buffers became processed in one loop, the following loop, we would have 
					 * at least that_many-1 buffers_processed (plus possible new processed).
					 * If we tried to just remove 1 buffer from our queue, we would be incorrect
					 * because we would not actually reflect the current playing buffer.
					 * So the solution seems to be to make sure our queue is the same size
					 * as the number of buffers_queued-buffers_processed, and return the head of our queue
					 * as the current playing buffer.
					 */
					/* Also, we have a corner case. When we first start playing or if we have 
					 * a buffer underrun, we have not done a data callback.
					 * In this case, we need to see if there is any new data in our queue
					 * and if so, launch that data callback.
					 */
					/* Warning, this code risks the possibility of no data callback being fired if 
					 * the system is really late (or skipped buffers).
					 */
					
					/* First, let's syncronize our queue with the OpenAL queue */
					#if 0
					fprintf(stderr, "inside, Buffers processed=%d, Buffers queued=%d, my queue=%d\n",
							buffers_processed, buffers_still_queued, my_queue_size);
					#endif			
					if(my_queue_size > (buffers_still_queued - buffers_processed))
					{
						is_out_of_sync = 1;
						for(k=0; k<(my_queue_size - (buffers_still_queued - buffers_processed)); k++)
						{
							queue_ret_flag = CircularQueueUnsignedInt_PopFront(
								ALmixer_Channel_List[i].almixer_data->circular_buffer_queue);
							if(0 == queue_ret_flag)
							{
								fprintf(stderr, "53 Error popping queue\n");
							}
						}						
					}
					my_queue_size = CircularQueueUnsignedInt_Size(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue);
					/* We have several possibilities we need to handle:
					 * 1) We are in an initial state or underrun and need to do a data callback on the head.
					 * 2) We were out of sync and need to do a new data callback on the new head.
					 * 3) We were not out of sync but just had left over processed buffers which caused us to
					 * fall in this block of code. (Don't do anything.)
					 */
					if( (0 == ALmixer_Channel_List[i].almixer_data->current_buffer) || (1 == is_out_of_sync) )
					{
						if(my_queue_size > 0)
						{
							current_buffer_id = CircularQueueUnsignedInt_Front(
								ALmixer_Channel_List[i].almixer_data->circular_buffer_queue);
							if(0 == current_buffer_id)
							{
								fprintf(stderr, "53a Internal Error, current_buffer_id=0 when it shouldn't be 0\n");
							}
							/*
							else
							{
								fprintf(stderr, "Queue in processed check, after pop\n");
								CircularQueueUnsignedInt_Print(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue);
							}
							 */
							ALmixer_Channel_List[i].almixer_data->current_buffer 
								= (ALuint)current_buffer_id;
							
							#if 0
							/* Remove me...only for checking...doesn't work on Nvidia */
							{
								ALuint real_id;
								alGetSourcei(
										 ALmixer_Channel_List[i].alsource,
										 AL_BUFFER, &real_id
										 );
								alGetError();
								fprintf(stderr, "Callback fired on data buffer=%d, real_id shoud be=%d\n", current_buffer_id, real_id);
							}
							#endif
							Invoke_Streamed_Channel_Data_Callback(i, ALmixer_Channel_List[i].almixer_data, current_buffer_id);
						}
						else
						{
							fprintf(stderr, "53b, Notice/Warning:, OpenAL queue has been depleted.\n");
							/* In this case, we might either be in an underrun or finished with playback */
							ALmixer_Channel_List[i].almixer_data->current_buffer = 0;
						}
					}
				}
#endif
					
				
				
		/* Just a test - remove 
				if( ALmixer_Channel_List[i].loops > 0)
				{
					fprintf(stderr, ">>>>>>>>>>>>>>>Loops = %d\n", 
						ALmixer_Channel_List[i].loops);
				}
		*/
#if 0
				fprintf(stderr, "Buffers processed = %d\n", buffers_processed);
				fprintf(stderr, "Buffers queued= %d\n", buffers_still_queued);
#endif
				/* We've used up a buffer so we need to unqueue and replace */
				/* Okay, it gets more complicated here:
				 * We need to Queue more data 
				 * if buffers_processed > 0  or 
				 * if num_of_buffers_in_use < NUMBER_OF_QUEUE_BUFFERS
				 * but we don't do this if at EOF,
				 * except when there is looping
				 */
				/* For this to work, we must rely on EVERYTHING
				 * else to unset the EOF if there is looping.
				 * Remember, even Play() must do this
				 */
				
				/* If not EOF, then we are still playing.
				 * Inside, we might find num_of_buffers < NUM...QUEUE_BUF..
				 * or buffers_process > 0 
				 * in which case we queue up.
				 * We also might find no buffers we need to fill,
				 * in which case we just keep going
				 */
				if( ! ALmixer_Channel_List[i].almixer_data->eof)
				{
					Uint32 bytes_returned;
					/* We have a priority. We first must assign
					 * unused buffers in reserve. If there is nothing
					 * left, then we may unqueue buffers. We can't
					 * do it the other way around because we will
					 * lose the pointer to the unqueued buffer 
					 */
					if(ALmixer_Channel_List[i].almixer_data->num_buffers_in_use 
						< ALmixer_Channel_List[i].almixer_data->max_queue_buffers) 
					{		
						/*
						fprintf(stderr, "Getting more data in NOT_EOF and num_buffers_in_use (%d) < max_queue (%d)\n", 
								ALmixer_Channel_List[i].almixer_data->num_buffers_in_use,
								ALmixer_Channel_List[i].almixer_data->max_queue_buffers);
						*/
						/* Going to add an unused packet.
						 * Grab next packet */
						bytes_returned = GetMoreData(
							ALmixer_Channel_List[i].almixer_data,
							ALmixer_Channel_List[i].almixer_data->buffer[
								ALmixer_Channel_List[i].almixer_data->num_buffers_in_use]
							);
					}
					/* For processed > 0 */
					else if(buffers_processed > 0)
					{
						/* Unqueue only 1 buffer for now.
						 * If there are more than one,
						 * let the next Update pass deal with it
						 * so we don't stall the program for too long.
						 */
#if 0
				fprintf(stderr, "About to Unqueue, Buffers processed = %d\n", buffers_processed);
				fprintf(stderr, "Buffers queued= %d\n", buffers_still_queued);
				fprintf(stderr, "Unqueuing a buffer\n");
#endif
						alSourceUnqueueBuffers(
							ALmixer_Channel_List[i].alsource,
							1, &unqueued_buffer_id
						);
						if((error = alGetError()) != AL_NO_ERROR)
						{
				fprintf(stderr, "Error with unqueue: %s",
							aluGetErrorString(error));
							ALmixer_SetError("Unqueue buffer failed: %s",
								aluGetErrorString(error) );
							error_flag--;
						}
/*
						fprintf(stderr, "Right after unqueue...");
						PrintQueueStatus(ALmixer_Channel_List[i].alsource);
						fprintf(stderr, "Getting more data for NOT_EOF, max_buffers filled\n");
*/
						/* Grab unqueued packet */
						bytes_returned = GetMoreData(
							ALmixer_Channel_List[i].almixer_data,
							unqueued_buffer_id);
					}
					/* We are still streaming, but currently
					 * don't need to fill any buffers */
					else
					{
						/* Might want to check state */
						/* In case the playback stopped,
						 * we need to resume */
						alGetSourcei(
							ALmixer_Channel_List[i].alsource,
							AL_SOURCE_STATE, &state
						);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "54Testing error: %s\n",
			aluGetErrorString(error));				
	}
						if(AL_STOPPED == state)
						{
							/* Resuming in not eof, but nothing to buffer */
							alSourcePlay(ALmixer_Channel_List[i].alsource);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "55Testing error: %s\n",
			aluGetErrorString(error));				
	}
						}
						/* Let's escape to the next loop.
						 * All code below this point is for queuing up 
						 */
						/*
		fprintf(stderr, "Entry: Nothing to do...continue\n\n");
		*/				
						continue;
					}
					/* We now know we have to fill an available
					 * buffer.
					 */
					
					/* In the previous branch, we just grabbed more data.
					 * Let's check it to make sure it's okay,
					 * and then queue it up
					 */
					/* This check doesn't work anymore because it is now Uint32 */
				#if 0
					if(-1 == bytes_returned)
					{
						/* Problem occurred...not sure what to do */
						/* Go to next loop? */
						error_flag--;
						/* Set the eof flag to force a quit so 
						 * we don't get stuck in an infinite loop
						 */
						ALmixer_Channel_List[i].almixer_data->eof = 1;
						continue;
					}
				#endif
					/* This is a special case where we've run
					 * out of data. We should check for loops
					 * and get more data. If there is no loop,
					 * then do nothing and wait for future
					 * update passes to handle the EOF.
					 * The advantage of handling the loop here 
					 * instead of waiting for play to stop is
					 * that we should be able to keep the buffer
					 * filled.
					 */
				#if 0
					else if(0 == bytes_returned)
				#endif
					if(0 == bytes_returned)
					{
				fprintf(stderr, "We got 0 bytes from reading. Checking for loops\n");
						/* Check for loops */
						if( ALmixer_Channel_List[i].loops != 0 )
						{
							/* We have to loop, so rewind
							 * and fetch more data 
							 */
				fprintf(stderr, "Rewinding data\n");
							if(0 == Sound_Rewind(
								ALmixer_Channel_List[i].almixer_data->sample))
							{
				fprintf(stderr, "Rewinding failed\n");
								ALmixer_SetError( Sound_GetError() );
								ALmixer_Channel_List[i].loops = 0;
								error_flag--;
								/* We'll continue on because we do have some valid data */
								continue;
							}
							/* Remember to reset the data->eof flag */
							ALmixer_Channel_List[i].almixer_data->eof = 0;
							if(ALmixer_Channel_List[i].loops > 0)
							{
								ALmixer_Channel_List[i].loops--;
							}
							/* Try grabbing another packet now.
							 * Since we may have already unqueued a
							 * buffer, we don't want to lose it.
							 */
							if(ALmixer_Channel_List[i].almixer_data->num_buffers_in_use 
								< ALmixer_Channel_List[i].almixer_data->max_queue_buffers) 
							{
				fprintf(stderr, "We got %d bytes from reading loop. Filling unused packet\n", bytes_returned);
								/* Grab next packet */
								bytes_returned = GetMoreData(
									ALmixer_Channel_List[i].almixer_data,
										ALmixer_Channel_List[i].almixer_data->buffer[
										ALmixer_Channel_List[i].almixer_data->num_buffers_in_use]
								);
				fprintf(stderr, "We reread %d bytes into unused packet\n", bytes_returned);
							}
							/* Refilling unqueued packet */
							else
							{
				fprintf(stderr, "We got %d bytes from reading loop. Filling unqueued packet\n", bytes_returned);
								/* Grab next packet */
								bytes_returned = GetMoreData(
									ALmixer_Channel_List[i].almixer_data,
									unqueued_buffer_id);
				fprintf(stderr, "We reread %d bytes into unqueued packet\n", bytes_returned);
							}	
							/* Another error check */
							/*
							if(bytes_returned <= 0)
							*/
							if(0 == bytes_returned)
							{
		fprintf(stderr, "??????????ERROR\n");
								ALmixer_SetError("Could not loop because after rewind, no data could be retrieved");
								/* Problem occurred...not sure what to do */
								/* Go to next loop? */
								error_flag--;
								/* Set the eof flag to force a quit so 
								 * we don't get stuck in an infinite loop
								 */
								ALmixer_Channel_List[i].almixer_data->eof = 1;
								continue;
							}	
							/* We made it to the end. We still need 
							 * to BufferData, so let this branch
							 * fall into the next piece of 
							 * code below which will handle that 
							 */

							
						} /* END loop check */
						else
						{
							/* No more loops to do. 
							 * EOF flag should be set.
							 * Just go to next loop and
							 * let things be handled correctly
							 * in future update calls
							 */
						fprintf(stderr, "SHOULD BE EOF\n");
							
							PrintQueueStatus(ALmixer_Channel_List[i].alsource);
							
							continue;
						}
					} /* END if bytes_returned == 0 */
/********* Possible trouble point. I might be queueing empty buffers on the mac. 
 * This check doesn't say if the buffer is valid. Only the EOF assumption is a clue at this point 
 */
					/* Fall here */
					/* Everything is normal. We aren't
					 * at an EOF, but need to simply
					 * queue more data. The data is already checked for good, 
					 * so queue it up */
					if(ALmixer_Channel_List[i].almixer_data->num_buffers_in_use 
						< ALmixer_Channel_List[i].almixer_data->max_queue_buffers) 
					{
						/* Keep count of how many buffers we have 
						 * to queue so we can return the value 
						 */
						retval++;
						fprintf(stderr, "NOT_EOF???, about to Queue more data for num_buffers (%d) < max_queue (%d)\n",
								ALmixer_Channel_List[i].almixer_data->num_buffers_in_use,
								ALmixer_Channel_List[i].almixer_data->max_queue_buffers);
								
						alSourceQueueBuffers(
							ALmixer_Channel_List[i].alsource,
							1, 
							&ALmixer_Channel_List[i].almixer_data->buffer[ 
								ALmixer_Channel_List[i].almixer_data->num_buffers_in_use]
						);
						if((error = alGetError()) != AL_NO_ERROR)
						{
							fprintf(stderr, "56Testing error: %s\n",
								aluGetErrorString(error));				
						}
						/* This is part of the hideous Nvidia workaround. In order to figure out
						 * which buffer to show during callbacks (for things like
						 * o-scopes), I must keep a copy of the buffers that are queued in my own
						 * data structure. This code will be called only if
						 * "access_data" was set, indicated by whether the queue is NULL.
						 */
						if(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue != NULL)
						{
							Uint32 queue_ret_flag;
							queue_ret_flag = CircularQueueUnsignedInt_PushBack(
								ALmixer_Channel_List[i].almixer_data->circular_buffer_queue, 
								ALmixer_Channel_List[i].almixer_data->buffer[ALmixer_Channel_List[i].almixer_data->num_buffers_in_use]
							);	
							if(0 == queue_ret_flag)
							{
								fprintf(stderr, "56aSerious internal error: CircularQueue could not push into queue.\n");
								ALmixer_SetError("Serious internal error: CircularQueue failed to push into queue");
							}
							/*
							else
							{
								CircularQueueUnsignedInt_Print(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue);
							}
							 */
						}
					}
					/* for processed > 0 */
					else
					{
						/* Keep count of how many buffers we have 
						 * to queue so we can return the value 
						 */
						retval++;
/*
						fprintf(stderr, "NOT_EOF, about to Queue more data for filled max_queue (%d)\n",
								ALmixer_Channel_List[i].almixer_data->max_queue_buffers);
*/
						alSourceQueueBuffers(
							ALmixer_Channel_List[i].alsource,
							1, &unqueued_buffer_id);
						if((error = alGetError()) != AL_NO_ERROR)
						{
							ALmixer_SetError("Could not QueueBuffer: %s",
								aluGetErrorString(error) );
							error_flag--;
							continue;
						}
						/* This is part of the hideous Nvidia workaround. In order to figure out
						 * which buffer to show during callbacks (for things like
						 * o-scopes), I must keep a copy of the buffers that are queued in my own
						 * data structure. This code will be called only if
						 * "access_data" was set, indicated by whether the queue is NULL.
						 */
						if(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue != NULL)
						{
							Uint32 queue_ret_flag;
							queue_ret_flag = CircularQueueUnsignedInt_PushBack(
								ALmixer_Channel_List[i].almixer_data->circular_buffer_queue, 
								unqueued_buffer_id
							);
							if(0 == queue_ret_flag)
							{
								fprintf(stderr, "56bSerious internal error: CircularQueue could not push into queue.\n");
								ALmixer_SetError("Serious internal error: CircularQueue failed to push into queue");
							}
							/*
							else
							{
								CircularQueueUnsignedInt_Print(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue);
							}
							 */
						}
					}
					/* If we used an available buffer queue,
					 * then we need to update the number of them in use
					 */
					if(ALmixer_Channel_List[i].almixer_data->num_buffers_in_use 
						< ALmixer_Channel_List[i].almixer_data->max_queue_buffers) 
					{
						/* Increment the number of buffers in use */
						ALmixer_Channel_List[i].almixer_data->num_buffers_in_use++;
					}
					/* Might want to check state */
					/* In case the playback stopped,
					 * we need to resume */
					alGetSourcei(
						ALmixer_Channel_List[i].alsource,
						AL_SOURCE_STATE, &state
					);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "57Testing error: %s\n",
			aluGetErrorString(error));				
	}
					if(AL_STOPPED == state)
					{
						fprintf(stderr, "Resuming in not eof\n");
						alSourcePlay(ALmixer_Channel_List[i].alsource);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "58Testing error: %s\n",
			aluGetErrorString(error));				
	}
					}
					continue;
				} /* END if( ! eof) */
				/* We have hit EOF in the SDL_Sound sample and there
				 * are no more loops. However, there may still be
				 * buffers in the OpenAL queue which still need to
				 * be played out. The following body of code will
				 * determine if play is still happening or 
				 * initiate the stop/cleanup sequenece.
				 */
				else
				{
					/* Let's continue to remove the used up
					 * buffers as they come in. */
					if(buffers_processed > 0)
					{
						Sint32 temp_count;
						/* Do as a for-loop because I don't want
						 * to have to create an array for the 
						 * unqueued_buffer_id's
						 */
						for(temp_count=0; temp_count<buffers_processed; temp_count++)
						{
							fprintf(stderr, "unqueuing remainder, %d\n", temp_count);
							alSourceUnqueueBuffers(
								ALmixer_Channel_List[i].alsource,
								1, &unqueued_buffer_id
							);
							if((error = alGetError()) != AL_NO_ERROR)
							{
								fprintf(stderr, "59Testing error: %s\n",
										aluGetErrorString(error));				
							}
						}
						fprintf(stderr, "done unqueuing remainder for this loop, %d\n", temp_count);
						
						/* Need to update counts since we removed everything. 
						 * If we don't update the counts here, we end up in the
						 *	"Shouldn't be here section, but maybe it's okay due to race conditions"
						 */
						alGetSourcei(
									 ALmixer_Channel_List[i].alsource,
									 AL_BUFFERS_QUEUED, &buffers_still_queued
									 );
						if((error = alGetError()) != AL_NO_ERROR)
						{
							fprintf(stderr, "5100Testing error: %s\n",
									aluGetErrorString(error));				
						}
						/* Get the number of buffers processed
							* so we know if we need to refill 
							*/
						alGetSourcei(
									 ALmixer_Channel_List[i].alsource,
									 AL_BUFFERS_PROCESSED, &buffers_processed
									 );
						if((error = alGetError()) != AL_NO_ERROR)
						{
							fprintf(stderr, "5200Testing error: %s\n",
									aluGetErrorString(error));				
						}
					}


					/* Else if buffers_processed == 0
					 * and buffers_still_queued == 0.
					 * then we check to see if the source
					 * is still playing. Quit if stopped
					 * We shouldn't need to worry about
					 * looping because that should have
					 * been handled above.
					 */
					if(0 == buffers_still_queued)
					{
						/* Make sure playback has stopped before
						 * we shutdown.
						 */
						alGetSourcei(
							ALmixer_Channel_List[i].alsource,
							AL_SOURCE_STATE, &state
						);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "60Testing error: %s\n",
			aluGetErrorString(error));				
	}
						if(AL_STOPPED == state)
						{
							ALmixer_Channel_List[i].almixer_data->num_buffers_in_use  = 0;
							/* Playback has ended. 
							 * Loop if necessary, or launch callback
							 * and clear channel (or clear channel and
							 * then launch callback?)
							 */
							Clean_Channel(i);
							/* Subtract counter */
							Is_Playing_global--;
	
							/* Launch callback */
							Invoke_Channel_Done_Callback(i);
	
							/* We're done for this loop.
							 * Go to next channel 
							 */
							continue;
						}
					} /* End end-playback */
					else
					{
						/* Need to run out buffer */
			#if 1
						/* Might want to check state */
						/* In case the playback stopped,
						 * we need to resume */
						alGetSourcei(
							ALmixer_Channel_List[i].alsource,
							AL_SOURCE_STATE, &state
						);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "61Testing error: %s\n",
			aluGetErrorString(error));				
	}
						if(AL_STOPPED == state)
						{
		fprintf(stderr, "Shouldn't be here. %d Buffers still in queue, but play stopped. This might be correct though because race conditions could have caused the STOP to happen right after our other tests...Checking queue status...\n", buffers_still_queued);

							PrintQueueStatus(ALmixer_Channel_List[i].alsource);
							
							/* Rather than force unqueuing the buffer, let's see if
							* setting the buffer to none works (the OpenAL 1.0 
							* Reference Annotation suggests this should work).								 
							*/
							alSourcei(ALmixer_Channel_List[i].alsource,
									  AL_BUFFER, AL_NONE); 
/*							
							PrintQueueStatus(ALmixer_Channel_List[i].alsource);
*/
							/* This doesn't work because in some cases, I think
							 * it causes the sound to be replayed
							 */
							/*
							fprintf(stderr, "Resuming in eof (trying to run out buffers\n");
							alSourcePlay(ALmixer_Channel_List[i].alsource);
		*/
						}
			#endif
					} /* End trap section */
				} /* End POST-EOF use-up buffer section */
			} /* END Streamed section */
		} /* END channel in use */
	} /* END for-loop for each channel */

#ifdef ENABLE_ALMIXER_ALC_SYNC	
	alcProcessContext(alcGetCurrentContext());
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "62Testing error: %s\n",
			aluGetErrorString(error));				
	}
#endif

	SDL_UnlockMutex(simple_lock);
	/* Return the number of errors */
	if(error_flag < 0) 
	{
		return error_flag;
	}
	/* Return the number of buffers that were queued */
	return retval;
}

#ifdef ENABLE_PARANOID_SIGNEDNESS_CHECK
/* This is only here so we can call SDL_OpenAudio() */
static void my_dummy_audio_callback(void* userdata, Uint8* stream, int len)
{
}
#endif




#ifdef ENABLE_ALMIXER_THREADS
/* We might need threads. We
 * must constantly poll OpenAL to find out
 * if sound is being streamed, if play has 
 * ended, etc. Without threads, this must
 * be explicitly done by the user.
 * We could try to do it for them if we 
 * finish the threads.
 */

static int Stream_Data_Thread_Callback(void* data)
{
	Sint32 retval;
	
	while(ALmixer_Initialized)
	{
		retval = Update_ALmixer(data);
		/* 0 means that nothing needed updating and 
		 * the function returned quickly
		 */
		if(0 == retval)
		{
			/* Let's be nice and make the thread sleep since
			 * little work was done in update
			 */
			/* Make sure times are multiples of 10
			 * for optimal performance and accuracy in Linux
			 */
			SDL_Delay(10);
		}
		else
		{
			/* should I also be sleeping/yielding here? */
			SDL_Delay(0);
		}
	}
fprintf(stderr, "Thread is closing\n");
	return 0;
}
#endif /* End of ENABLE_ALMIXER_THREADS */


/* Using -1 on error and 0 on success to follow SDL/SDL_mixer conventions,
 * though I actually prefer 0/1 conventions (SDL_Sound/OpenAL/GL).
 * Warning: SDL_QuitSubSystem(SDL_INIT_AUDIO) is called which
 * means the SDL audio system will be disabled. It will not
 * be restored (in case SDL is not actually being used) so
 * the user will need to restart it if they need it after
 * OpenAL shuts down.
 */
Sint32 ALmixer_Init(Uint32 frequency, Sint32 num_sources, Uint32 refresh)
{
	ALCdevice* dev;
	ALCcontext* context;
	Sint32 i;
	ALenum error;
	ALuint* source;

#ifdef USING_LOKI_AL_DIST
	/* The Loki dist requires that I set both the 
	 * device and context frequency values separately
	 */
	/* Hope this won't overflow */
	char device_string[256];
#endif
	
	/* (Venting frustration) Damn it! Nobody bothered
	 * documenting how you're supposed to use an attribute
	 * list. In fact, the not even the Loki test program
	 * writers seem to know because they use it inconsistently.
	 * For example, how do you terminate that attribute list?
	 * The Loki test code does it 3 different ways. They 
	 * set the last value to 0, or they set it to ALC_INVALID, 
	 * or they set two final values: ALC_INVALID, 0
	 * In Loki, 0 and ALC_INVALID happen to be the same,
	 * but with Creative Labs ALC_INVALID is -1. 
	 * So something's going to break. Loki's source
	 * code says to terminate with ALC_INVALID. But I 
	 * don't know if that's really true, or it happens 
	 * to be a coinicidence because it's defined to 0.
	 * Creative provides no source code, so I can't look at how 
	 * they terminate it. 
	 * So this is really, really ticking me off...
	 * For now, I'm going to use ALC_INVALID.
	 * (Update...after further review of the API spec,
	 * it seems that a NULL terminated string is the correct
	 * termination value to use, so 0 it is.)
	 */
#if 0
	ALint attrlist[] = { 
		ALC_FREQUENCY, ALMIXER_DEFAULT_FREQUENCY,
		/* Don't know anything about these values.
		 * Trust defaults? */
		/* Supposed to be the refresh rate in Hz.
		 * I think 15-120 are supposed to be good 
		 * values. Though I haven't gotten any effect except
		 * for one strange instance on a Mac. But it was 
		 * unrepeatable.
		 */
	#if 0
		ALC_REFRESH, 15,
	#endif	
		/* Sync requires a alcProcessContext() call
		 * for every cycle. By default, this is
		 * not used and the value is AL_FALSE
		 * because it will probably perform
		 * pretty badly for me.
		 */
#ifdef ENABLE_ALMIXER_ALC_SYNC	
		ALC_SYNC, AL_TRUE,
#else
		ALC_SYNC, AL_FALSE,
#endif
		/* Looking at the API spec, it implies
		 * that the list be a NULL terminated string
		 * so it's probably not safe to use ALC_INVALID
		*/
		/*
		ALC_INVALID };
		*/
		'\0'};
#endif
	/* Redo: I'm going to allow ALC_REFRESH to be set.
	 * However, if no value is specified, I don't
	 * want it in the list so I can get the OpenAL defaults
	 */
	ALint attrlist[7];

#ifdef ENABLE_PARANOID_SIGNEDNESS_CHECK
	/* More problems: I'm getting bit by endian/signedness issues on
	 * different platforms. I can find the endianess easily enough,
	 * but I don't know how to determine what the correct signedness
	 * is (if such a thing exists). I do know that if I try using
	 * unsigned on OSX with an originally signed sample, I get 
	 * distortion. However, I don't have any native unsigned samples
	 * to test. But I'm assuming that the platform must be in the 
	 * correct signedness no matter what.
	 * I can either assume everybody is signed, or I can try to 
	 * determine the value. If I try to determine the values,
	 * I think my only ability to figure it out will be to open
	 * SDL_Audio, and read what the obtained settings were.
	 * Then shutdown everything. However, I don't even know how 
	 * reliable this is.
	 * Update: I think I resolved the issues...forgot to update 
	 * these comments when it happened. I should check the revision control
	 * log... Anyway, I think the issue was partly related to me not 
	 * doing something correctly with the AudioInfo or some kind 
	 * of stupid endian bug in my code, and weirdness ensued. Looking at the
	 * revision control, I think I might have assumed that SDL_Sound would 
	 * do the right thing with a NULL AudioInfo, but I was incorrect,
	 * and had to fill one out myself.
	 */
	SDL_AudioSpec desired;
	SDL_AudioSpec obtained;
#endif




	/* Make sure ALmixer isn't already initialized */
	if(ALmixer_Initialized)
	{
		return -1;
	}
#ifdef USING_LOKI_AL_DIST
fprintf(stderr, "Found Loki dist\n");
#elif defined(USING_CREATIVE_AL_DIST)
fprintf(stderr, "Found Creative dist\n");

#elif defined(USING_NVIDIA_AL_DIST)
fprintf(stderr, "Found Nvidia dist\n");
#endif

	/* Set the defaults */
	attrlist[0] = ALC_FREQUENCY;
	attrlist[1] = ALMIXER_DEFAULT_FREQUENCY;
	attrlist[2] = ALC_SYNC;
#ifdef ENABLE_ALMIXER_ALC_SYNC	
	attrlist[3] = ALC_TRUE;
#else
	attrlist[3] = ALC_FALSE;
#endif
	
	/* Set frequency value if it is not 0 */
	if(0 != frequency)
	{
		attrlist[1] = (ALint)frequency;
	}

	/* If the user specifies a refresh value,
	 * make room for it 
	 */
	if(0 != refresh)
	{
		attrlist[4] = (ALint)ALC_REFRESH;
		attrlist[5] = refresh;
		attrlist[6] = '\0';
	}
	/* Terminate the list without any refresh values */
	else
	{
		attrlist[4] = '\0';
	}
			
	/* It looks like OpenAL won't let us ask it what 
	 * the set frequency is, so we need to save our
	 * own copy. Yuck.
	 * Update: J. Valenzuela just updated the Loki 
	 * dist (2003/01/02) to handle this.
	 * The demo is in testattrib.c. However, this
	 * looks kind of cumbersome to parse, and I've
	 * already put this in my code, so I guess I'll 
	 * leave it for now.
	 */
	ALmixer_Frequency_global = attrlist[1];
	
	/* Initialize SDL_Sound */
	if(! Sound_Init() )
	{
		ALmixer_SetError(Sound_GetError());
		return -1;
	}
#ifdef ENABLE_PARANOID_SIGNEDNESS_CHECK
	/* Here is the paranoid check that opens
	 * SDL audio in an attempt to find the correct 
	 * system values.
	 */
	/* Doesn't have to be the actual value I think
	 * (as long as it doesn't influence format, in 
	 * which case I'm probably screwed anyway because OpenAL
	 * may easily choose to do something else).
	 */
	desired.freq = 44100;
	desired.channels = 2;
	desired.format = AUDIO_S16SYS;
	desired.callback = my_dummy_audio_callback;
	if(SDL_OpenAudio(&desired, &obtained) >= 0)
	{
		SIGN_TYPE_16BIT_FORMAT = obtained.format;
		/* Now to get really paranoid, we should probably
		 * also assume that the 8bit format is also the
		 * same sign type and set that value
		 */
		if(AUDIO_S16SYS == obtained.format)
		{
			SIGN_TYPE_8BIT_FORMAT = AUDIO_S8;
		}
		/* Should be AUDIO_U16SYS */
		else
		{
			SIGN_TYPE_8BIT_FORMAT = AUDIO_U8;
		}
		SDL_CloseAudio();
fprintf(stderr, "Obtained format = %d", obtained.format);
	}
	else
	{
		/* Well, I guess I'm in trouble. I guess it's my best guess
		 */
		SIGN_TYPE_16_BIT_FORMAT = AUDIO_S16SYS;
		SIGN_TYPE_8_BIT_FORMAT = AUDIO_S8;
	}
#endif
	/* Weirdness: It seems that SDL_Init(SDL_INIT_AUDIO)
	 * causes OpenAL and SMPEG to conflict. For some reason
	 * if SDL_Init on audio is active, then all the SMPEG
	 * decoded sound comes out silent. Unfortunately,
	 * Sound_Init() invokes SDL_Init on audio. I'm
	 * not sure why it actually needs it...
	 * But we'll attempt to disable it here after the
	 * SDL_Sound::Init call and hope it doesn't break SDL_Sound.
	 */
	SDL_QuitSubSystem(SDL_INIT_AUDIO);
	
	/* I'm told NULL will call the default string
	 * and hopefully do the right thing for each platform 
	 */
	/*
	dev = alcOpenDevice( NULL );
	*/
	/* Now I'm told I need to set both the device and context
	 * to have the same sampling rate, so I must pass a string
	 * to OpenDevice(). I don't know how portable these strings are.
	 * I don't even know if the format for strings is 
	 * compatible
	 * From the testattrib.c in the Loki test section
	 * dev = alcOpenDevice(  (const ALubyte *) "'((sampling-rate 22050))" );
	 */	

#ifdef USING_LOKI_AL_DIST
	sprintf(device_string, "'((sampling-rate %d))", attrlist[1]);
	dev = alcOpenDevice(  (const ALubyte *) device_string );
#else
	dev = alcOpenDevice( NULL );
#endif
	fprintf(stderr,"sampling-rate is %d\n", attrlist[1]);
	if(NULL == dev)
	{
		ALmixer_SetError("Cannot open sound device for OpenAL");
		return -1;
	}
	
	context = alcCreateContext(dev, attrlist);
	if(NULL == context)
	{
		ALmixer_SetError("Cannot create a context OpenAL");
		alcCloseDevice(dev);
		return -1;
	}
	fprintf(stderr, "Context checking...\n");


	/* Hmmm, OSX is returning 1 on alcMakeContextCurrent,
	 * but ALC_NO_ERROR is defined to ALC_FALSE.
	 * According to Garin Hiebert, this is actually an inconsistency
	 * in the Loki version. The function should return a boolean.
	 * instead of ALC_NO_ERROR. Garin suggested I check via
	 * alcGetError().
	 */
	/* clear the error */
	alcGetError(dev);
	alcMakeContextCurrent(context);
	
	error = alcGetError(dev);
	if( (ALC_NO_ERROR != error) )
	{
		ALmixer_SetError("Could not MakeContextCurrent");
		alcDestroyContext(context);
		alcCloseDevice(dev);
		return -1;
	}
	
	
#if 0
	/* OSX is failing on alcMakeContextCurrent(). Try checking it first? */
	if(alcGetCurrentContext() != context)
	{
		/* Hmmm, OSX is returning 1 on alcMakeContextCurrent,
		 * but ALC_NO_ERROR is defined to ALC_FALSE.
		 * I think this is a bug in the OpenAL implementation.
		 */
		fprintf(stderr,"alcMakeContextCurrent returns %d\n", alcMakeContextCurrent(context));
				
		fprintf(stderr, "Making context current\n");
#ifndef __APPLE__
		if(alcMakeContextCurrent(context) != ALC_NO_ERROR)
#else
		if(!alcMakeContextCurrent(context))
#endif			
		{
			ALmixer_SetError("Could not MakeContextCurrent");
			alcDestroyContext(context);
			alcCloseDevice(dev);
			return -1;
		}
	}
#endif


/* #endif */
	fprintf(stderr, "done Context\n");
	/* Saw this in the README with the OS X OpenAL distribution.
	 * It looked interesting and simple, so I thought I might
	 * try it out.
	 * ***** ALC_CONVERT_DATA_UPON_LOADING
	 * This extension allows the caller to tell OpenAL to preconvert to the native Core
	 * Audio format, the audio data passed to the 
	 * library with the alBufferData() call. Preconverting the audio data, reduces CPU 
	 * usage by removing an audio data conversion 
	 * (per source) at render timem at the expense of a larger memory footprint.
	 *
	 *	This feature is toggled on/off by using the alDisable() & alEnable() APIs. This 
	 *	setting will be applied to all subsequent 
	 *	calls to alBufferData().
	 */	
#ifdef __APPLE__
	alEnable(ALC_CONVERT_DATA_UPON_LOADING);
#endif
	

	
	
	ALmixer_Initialized = 1;

	if(num_sources <= 0)
	{
		Number_of_Channels_global = ALMIXER_DEFAULT_NUM_CHANNELS;
	}
	else
	{
		Number_of_Channels_global = num_sources;
	}
	Number_of_Reserve_Channels_global = 0;
	Is_Playing_global = 0;
	/* Set to Null in case system quit and was reinitialized */
	Channel_Done_Callback = NULL;
	Channel_Done_Callback_Userdata = NULL;
	Channel_Data_Callback = NULL;

	/* Allocate memory for the list of channels */
	ALmixer_Channel_List = (struct ALmixer_Channel*) malloc(Number_of_Channels_global * sizeof(struct ALmixer_Channel));
	if(NULL == ALmixer_Channel_List)
	{
		ALmixer_SetError("Out of Memory for Channel List");
		alcDestroyContext(context);
		alcCloseDevice(dev);
		ALmixer_Initialized = 0;
		Number_of_Channels_global = 0;
		return -1;
	}

	/* Allocate memory for the list of sources that map to the channels */
	Source_Map_List = (Source_Map*) malloc(Number_of_Channels_global * sizeof(Source_Map));
	if(NULL == Source_Map_List)
	{
		ALmixer_SetError("Out of Memory for Source Map List");
		free(ALmixer_Channel_List);
		alcDestroyContext(context);
		alcCloseDevice(dev);
		ALmixer_Initialized = 0;
		Number_of_Channels_global = 0;
		return -1;
	}

	/* Create array that will hold the sources */
	source = (ALuint*)malloc(Number_of_Channels_global * sizeof(ALuint));
	if(NULL == source)
	{
		ALmixer_SetError("Out of Memory for sources");
		free(Source_Map_List);
		free(ALmixer_Channel_List);
		alcDestroyContext(context);
		alcCloseDevice(dev);
		ALmixer_Initialized = 0;
		Number_of_Channels_global = 0;
		return -1;
	}	

	/* Clear the error state */
	alGetError();
	/* Generate the OpenAL sources */
	alGenSources(Number_of_Channels_global, source);
	if( (error=alGetError()) != AL_NO_ERROR)
	{
		ALmixer_SetError("Couldn't generate sources: %s\n", aluGetErrorString(error));
		free(ALmixer_Channel_List);
		free(Source_Map_List);
		alcDestroyContext(context);
		alcCloseDevice(dev);
		ALmixer_Initialized = 0;
		Number_of_Channels_global = 0;
		return -1;
	}
	
	/* Initialize each channel and associate one source to one channel */
	for(i=0; i<Number_of_Channels_global; i++)
	{
		if(0 == source[i])
		{
			fprintf(stderr, "SDL_ALmixer serious problem. This OpenAL implementation allowed 0 to be a valid source id which is in conflict with assumptions made in this library.\n");
		}
		
		Init_Channel(i);
		/* Keeping the source allocation out of the Init function
		 * in case I want to reuse the Init
		 * function for resetting data 
		 */
		ALmixer_Channel_List[i].alsource = source[i];
		/* Now also keep a copy of the source to channel mapping
		 * in case we need to look up a channel from the source
		 * instead of a source from a channel 
		 */
		Source_Map_List[i].source = source[i];
		Source_Map_List[i].channel = i;
		/* Clean the channel because there are some things that need to 
		 * be done that can't happen until the source is set
		 */
		Clean_Channel(i);
	}

	/* The Source_Map_List must be sorted by source for binary searches
	 */
	qsort(Source_Map_List, Number_of_Channels_global, sizeof(Source_Map), Compare_Source_Map);
	
	fprintf(stderr, "Sorted Source_Map_List is:\n");
	for(i=0; i<Number_of_Channels_global; i++)
	{
		fprintf(stderr, "Source: %d, Channel: %d\n", Source_Map_List[i].source, Source_Map_List[i].channel);
	}
	fprintf(stderr, "\n");
	
	simple_lock = SDL_CreateMutex();
	if(NULL == simple_lock)
	{
		/* SDL sets the error message already? */
		free(source);
		free(ALmixer_Channel_List);
		free(Source_Map_List);
		alcDestroyContext(context);
		alcCloseDevice(dev);
		ALmixer_Initialized = 0;
		Number_of_Channels_global = 0;
		return -1;
	}
		

#ifdef ENABLE_ALMIXER_THREADS
	Stream_Thread_global = SDL_CreateThread(Stream_Data_Thread_Callback, NULL);
	if(NULL == Stream_Thread_global)
	{
		/* SDL sets the error message already? */
		SDL_DestroyMutex(simple_lock);
		free(source);
		free(ALmixer_Channel_List);
		free(Source_Map_List);
		alcDestroyContext(context);
		alcCloseDevice(dev);
		ALmixer_Initialized = 0;
		Number_of_Channels_global = 0;
		return -1;
	}
		
	fprintf(stderr, "Using threads\n");
#endif /* End of ENABLE_ALMIXER_THREADS */

	/* We don't need this array any more because all the sources 
	 * are connected to channels
	 */
	free(source);
	return 0;
}

	
Sint32 ALmixer_Init_Context(Uint32 frequency, Uint32 refresh)
{
	ALCdevice* dev;
	ALCcontext* context;
	ALCenum error;

#ifdef USING_LOKI_AL_DIST
	/* The Loki dist requires that I set both the 
	 * device and context frequency values separately
	 */
	/* Hope this won't overflow */
	char device_string[256];
#endif
	
	/* (Venting frustration) Damn it! Nobody bothered
	 * documenting how you're supposed to use an attribute
	 * list. In fact, the not even the Loki test program
	 * writers seem to know because they use it inconsistently.
	 * For example, how do you terminate that attribute list?
	 * The Loki test code does it 3 different ways. They 
	 * set the last value to 0, or they set it to ALC_INVALID, 
	 * or they set two final values: ALC_INVALID, 0
	 * In Loki, 0 and ALC_INVALID happen to be the same,
	 * but with Creative Labs ALC_INVALID is -1. 
	 * So something's going to break. Loki's source
	 * code says to terminate with ALC_INVALID. But I 
	 * don't know if that's really true, or it happens 
	 * to be a coinicidence because it's defined to 0.
	 * Creative provides no source code, so I can't look at how 
	 * they terminate it. 
	 * So this is really, really ticking me off...
	 * For now, I'm going to use ALC_INVALID.
	 * (Update...after further review of the API spec,
	 * it seems that a NULL terminated string is the correct
	 * termination value to use, so 0 it is.)
	 */
#if 0
	ALint attrlist[] = { 
		ALC_FREQUENCY, ALMIXER_DEFAULT_FREQUENCY,
		/* Don't know anything about these values.
		 * Trust defaults? */
		/* Supposed to be the refresh rate in Hz.
		 * I think 15-120 are supposed to be good 
		 * values. Though I haven't gotten any effect except
		 * for one strange instance on a Mac. But it was 
		 * unrepeatable.
		 */
	#if 0
		ALC_REFRESH, 15,
	#endif	
		/* Sync requires a alcProcessContext() call
		 * for every cycle. By default, this is
		 * not used and the value is AL_FALSE
		 * because it will probably perform
		 * pretty badly for me.
		 */
#ifdef ENABLE_ALMIXER_ALC_SYNC	
		ALC_SYNC, AL_TRUE,
#else
		ALC_SYNC, AL_FALSE,
#endif
		/* Looking at the API spec, it implies
		 * that the list be a NULL terminated string
		 * so it's probably not safe to use ALC_INVALID
		*/
		/*
		ALC_INVALID };
		*/
		'\0'};
#endif
	/* Redo: I'm going to allow ALC_REFRESH to be set.
	 * However, if no value is specified, I don't
	 * want it in the list so I can get the OpenAL defaults
	 */
	ALint attrlist[7];

#ifdef ENABLE_PARANOID_SIGNEDNESS_CHECK
	/* More problems: I'm getting bit by endian/signedness issues on
	 * different platforms. I can find the endianess easily enough,
	 * but I don't know how to determine what the correct signedness
	 * is (if such a thing exists). I do know that if I try using
	 * unsigned on OSX with an originally signed sample, I get 
	 * distortion. However, I don't have any native unsigned samples
	 * to test. But I'm assuming that the platform must be in the 
	 * correct signedness no matter what.
	 * I can either assume everybody is signed, or I can try to 
	 * determine the value. If I try to determine the values,
	 * I think my only ability to figure it out will be to open
	 * SDL_Audio, and read what the obtained settings were.
	 * Then shutdown everything. However, I don't even know how 
	 * reliable this is.
	 * Update: I think I resolved the issues...forgot to update 
	 * these comments when it happened. I should check the revision control
	 * log... Anyway, I think the issue was partly related to me not 
	 * doing something correctly with the AudioInfo or some kind 
	 * of stupid endian bug in my code, and weirdness ensued. Looking at the
	 * revision control, I think I might have assumed that SDL_Sound would 
	 * do the right thing with a NULL AudioInfo, but I was incorrect,
	 * and had to fill one out myself.
	 */
	SDL_AudioSpec desired;
	SDL_AudioSpec obtained;
#endif




	/* Make sure ALmixer isn't already initialized */
	if(ALmixer_Initialized)
	{
		return -1;
	}
#ifdef USING_LOKI_AL_DIST
fprintf(stderr, "Found Loki dist\n");
#elif defined(USING_CREATIVE_AL_DIST)
fprintf(stderr, "Found Creative dist\n");

#elif defined(USING_NVIDIA_AL_DIST)
fprintf(stderr, "Found Nvidia dist\n");
#endif

	/* Set the defaults */
	attrlist[0] = ALC_FREQUENCY;
	attrlist[1] = ALMIXER_DEFAULT_FREQUENCY;
	attrlist[2] = ALC_SYNC;
#ifdef ENABLE_ALMIXER_ALC_SYNC	
	attrlist[3] = ALC_TRUE;
#else
	attrlist[3] = ALC_FALSE;
#endif
	
	/* Set frequency value if it is not 0 */
	if(0 != frequency)
	{
		attrlist[1] = (ALint)frequency;
	}

	/* If the user specifies a refresh value,
	 * make room for it 
	 */
	if(0 != refresh)
	{
		attrlist[4] = (ALint)ALC_REFRESH;
		attrlist[5] = refresh;
		attrlist[6] = '\0';
	}
	/* Terminate the list without any refresh values */
	else
	{
		attrlist[4] = '\0';
	}
			
	/* It looks like OpenAL won't let us ask it what 
	 * the set frequency is, so we need to save our
	 * own copy. Yuck.
	 * Update: J. Valenzuela just updated the Loki 
	 * dist (2003/01/02) to handle this.
	 * The demo is in testattrib.c. However, this
	 * looks kind of cumbersome to parse, and I've
	 * already put this in my code, so I guess I'll 
	 * leave it for now.
	 */
	ALmixer_Frequency_global = attrlist[1];
	
	/* Initialize SDL_Sound */
	if(! Sound_Init() )
	{
		ALmixer_SetError(Sound_GetError());
		return -1;
	}
#ifdef ENABLE_PARANOID_SIGNEDNESS_CHECK
	/* Here is the paranoid check that opens
	 * SDL audio in an attempt to find the correct 
	 * system values.
	 */
	/* Doesn't have to be the actual value I think
	 * (as long as it doesn't influence format, in 
	 * which case I'm probably screwed anyway because OpenAL
	 * may easily choose to do something else).
	 */
	desired.freq = 44100;
	desired.channels = 2;
	desired.format = AUDIO_S16SYS;
	desired.callback = my_dummy_audio_callback;
	if(SDL_OpenAudio(&desired, &obtained) >= 0)
	{
		SIGN_TYPE_16BIT_FORMAT = obtained.format;
		/* Now to get really paranoid, we should probably
		 * also assume that the 8bit format is also the
		 * same sign type and set that value
		 */
		if(AUDIO_S16SYS == obtained.format)
		{
			SIGN_TYPE_8BIT_FORMAT = AUDIO_S8;
		}
		/* Should be AUDIO_U16SYS */
		else
		{
			SIGN_TYPE_8BIT_FORMAT = AUDIO_U8;
		}
		SDL_CloseAudio();
fprintf(stderr, "Obtained format = %d", obtained.format);
	}
	else
	{
		/* Well, I guess I'm in trouble. I guess it's my best guess
		 */
		SIGN_TYPE_16_BIT_FORMAT = AUDIO_S16SYS;
		SIGN_TYPE_8_BIT_FORMAT = AUDIO_S8;
	}
#endif
	/* Weirdness: It seems that SDL_Init(SDL_INIT_AUDIO)
	 * causes OpenAL and SMPEG to conflict. For some reason
	 * if SDL_Init on audio is active, then all the SMPEG
	 * decoded sound comes out silent. Unfortunately,
	 * Sound_Init() invokes SDL_Init on audio. I'm
	 * not sure why it actually needs it...
	 * But we'll attempt to disable it here after the
	 * SDL_Sound::Init call and hope it doesn't break SDL_Sound.
	 */
	SDL_QuitSubSystem(SDL_INIT_AUDIO);
	
	/* I'm told NULL will call the default string
	 * and hopefully do the right thing for each platform 
	 */
	/*
	dev = alcOpenDevice( NULL );
	*/
	/* Now I'm told I need to set both the device and context
	 * to have the same sampling rate, so I must pass a string
	 * to OpenDevice(). I don't know how portable these strings are.
	 * I don't even know if the format for strings is 
	 * compatible
	 * From the testattrib.c in the Loki test section
	 * dev = alcOpenDevice(  (const ALubyte *) "'((sampling-rate 22050))" );
	 */	

#ifdef USING_LOKI_AL_DIST
	sprintf(device_string, "'((sampling-rate %d))", attrlist[1]);
	dev = alcOpenDevice(  (const ALubyte *) device_string );
#else
	dev = alcOpenDevice( NULL );
#endif
	fprintf(stderr,"sampling-rate is %d\n", attrlist[1]);
	if(NULL == dev)
	{
		ALmixer_SetError("Cannot open sound device for OpenAL");
		return -1;
	}
	
	context = alcCreateContext(dev, attrlist);
	if(NULL == context)
	{
		ALmixer_SetError("Cannot create a context OpenAL");
		alcCloseDevice(dev);
		return -1;
	}


	/* Hmmm, OSX is returning 1 on alcMakeContextCurrent,
	 * but ALC_NO_ERROR is defined to ALC_FALSE.
	 * According to Garin Hiebert, this is actually an inconsistency
	 * in the Loki version. The function should return a boolean.
	 * instead of ALC_NO_ERROR. Garin suggested I check via
	 * alcGetError().
	 */
	/* clear the error */
	alcGetError(dev);
	alcMakeContextCurrent(context);
	
	error = alcGetError(dev);
	if( (ALC_NO_ERROR != error) )
	{
		ALmixer_SetError("Could not MakeContextCurrent");
		alcDestroyContext(context);
		alcCloseDevice(dev);
		return -1;
	}


#if 0
	/* OSX is failing on alcMakeContextCurrent(). Try checking it first? */
	if(alcGetCurrentContext() != context)
	{
		/* Hmmm, OSX is returning 1 on alcMakeContextCurrent,
		 * but ALC_NO_ERROR is defined to ALC_FALSE.
		 * I think this is a bug in the OpenAL implementation.
		 */
		fprintf(stderr,"alcMakeContextCurrent returns %d\n", alcMakeContextCurrent(context));
				
		fprintf(stderr, "Making context current\n");
#ifndef __APPLE__
		if(alcMakeContextCurrent(context) != ALC_NO_ERROR)
#else
		if(!alcMakeContextCurrent(context))
#endif			
		{
			ALmixer_SetError("Could not MakeContextCurrent");
			alcDestroyContext(context);
			alcCloseDevice(dev);
			return -1;
		}

	}
#endif

	fprintf(stderr, "done Context\n");

	/* Saw this in the README with the OS X OpenAL distribution.
	 * It looked interesting and simple, so I thought I might
	 * try it out.
	 * ***** ALC_CONVERT_DATA_UPON_LOADING
	 * This extension allows the caller to tell OpenAL to preconvert to the native Core
	 * Audio format, the audio data passed to the 
	 * library with the alBufferData() call. Preconverting the audio data, reduces CPU 
	 * usage by removing an audio data conversion 
	 * (per source) at render timem at the expense of a larger memory footprint.
	 *
	 *	This feature is toggled on/off by using the alDisable() & alEnable() APIs. This 
	 *	setting will be applied to all subsequent 
	 *	calls to alBufferData().
	 */	
#ifdef __APPLE__
	alEnable(ALC_CONVERT_DATA_UPON_LOADING);
#endif
	
	return 0;
}

	
Sint32 ALmixer_Init_Mixer(Sint32 num_sources)
{	
	Sint32 i;
	ALenum error;
	ALuint* source;


	ALmixer_Initialized = 1;

	if(num_sources <= 0)
	{
		Number_of_Channels_global = ALMIXER_DEFAULT_NUM_CHANNELS;
	}
	else
	{
		Number_of_Channels_global = num_sources;
	}
	Number_of_Reserve_Channels_global = 0;
	Is_Playing_global = 0;
	/* Set to Null in case system quit and was reinitialized */
	Channel_Done_Callback = NULL;
	Channel_Done_Callback_Userdata = NULL;
	Channel_Data_Callback = NULL;

	/* Allocate memory for the list of channels */
	ALmixer_Channel_List = (struct ALmixer_Channel*) malloc(Number_of_Channels_global * sizeof(struct ALmixer_Channel));
	if(NULL == ALmixer_Channel_List)
	{
		ALmixer_SetError("Out of Memory for Channel List");
		ALmixer_Initialized = 0;
		Number_of_Channels_global = 0;
		return -1;
	}

	/* Allocate memory for the list of sources that map to the channels */
	Source_Map_List = (Source_Map*) malloc(Number_of_Channels_global * sizeof(Source_Map));
	if(NULL == Source_Map_List)
	{
		ALmixer_SetError("Out of Memory for Source Map List");
		free(ALmixer_Channel_List);
		ALmixer_Initialized = 0;
		Number_of_Channels_global = 0;
		return -1;
	}

	/* Create array that will hold the sources */
	source = (ALuint*)malloc(Number_of_Channels_global * sizeof(ALuint));
	if(NULL == source)
	{
		ALmixer_SetError("Out of Memory for sources");
		free(Source_Map_List);
		free(ALmixer_Channel_List);
		ALmixer_Initialized = 0;
		Number_of_Channels_global = 0;
		return -1;
	}	

	/* Clear the error state */
	alGetError();
	/* Generate the OpenAL sources */
	alGenSources(Number_of_Channels_global, source);
	if( (error=alGetError()) != AL_NO_ERROR)
	{
		ALmixer_SetError("Couldn't generate sources: %s\n", aluGetErrorString(error));
		free(ALmixer_Channel_List);
		free(Source_Map_List);
		ALmixer_Initialized = 0;
		Number_of_Channels_global = 0;
		return -1;
	}
	
	/* Initialize each channel and associate one source to one channel */
	for(i=0; i<Number_of_Channels_global; i++)
	{
		Init_Channel(i);
		/* Keeping the source allocation out of the Init function
		 * in case I want to reuse the Init
		 * function for resetting data 
		 */
		ALmixer_Channel_List[i].alsource = source[i];
		/* Now also keep a copy of the source to channel mapping
		 * in case we need to look up a channel from the source
		 * instead of a source from a channel 
		 */
		Source_Map_List[i].source = source[i];
		Source_Map_List[i].channel = i;
		/* Clean the channel because there are some things that need to 
		 * be done that can't happen until the source is set
		 */
		Clean_Channel(i);
	}

	/* The Source_Map_List must be sorted by source for binary searches
	 */
	qsort(Source_Map_List, Number_of_Channels_global, sizeof(Source_Map), Compare_Source_Map);
	
	fprintf(stderr, "Sorted Source_Map_List is:\n");
	for(i=0; i<Number_of_Channels_global; i++)
	{
		fprintf(stderr, "Source: %d, Channel: %d\n", Source_Map_List[i].source, Source_Map_List[i].channel);
	}
	fprintf(stderr, "\n");
	
	simple_lock = SDL_CreateMutex();
	if(NULL == simple_lock)
	{
		/* SDL sets the error message already? */
		free(source);
		free(ALmixer_Channel_List);
		free(Source_Map_List);
		ALmixer_Initialized = 0;
		Number_of_Channels_global = 0;
		return -1;
	}
		

#ifdef ENABLE_ALMIXER_THREADS
	Stream_Thread_global = SDL_CreateThread(Stream_Data_Thread_Callback, NULL);
	if(NULL == Stream_Thread_global)
	{
		/* SDL sets the error message already? */
		SDL_DestroyMutex(simple_lock);
		free(source);
		free(ALmixer_Channel_List);
		free(Source_Map_List);
		ALmixer_Initialized = 0;
		Number_of_Channels_global = 0;
		return -1;
	}
		
	fprintf(stderr, "Using threads\n");
#endif /* End of ENABLE_ALMIXER_THREADS */

	/* We don't need this array any more because all the sources 
	 * are connected to channels
	 */
	free(source);
	return 0;
}



/* Keep the return value void to allow easy use with
 * atexit()
 */
void ALmixer_Quit()
{
	ALCcontext* context;
	ALCdevice* dev;
	Sint32 i;
	
	if( ! ALmixer_Initialized)
	{
		return;
	}

	SDL_LockMutex(simple_lock);
	
	/* Shutdown everything before closing context */
	fprintf(stderr, "Halting channels\n");
	Internal_HaltChannel(-1);
	
	/* This flag will cause the thread to terminate */
	ALmixer_Initialized = 0;
	SDL_UnlockMutex(simple_lock);
#ifdef ENABLE_ALMIXER_THREADS
	fprintf(stderr, "Closing thread\n");
	SDL_WaitThread(Stream_Thread_global, NULL);
#endif

	fprintf(stderr, "Destroying mutex\n");
	SDL_DestroyMutex(simple_lock);

	fprintf(stderr, "Deleting OpenAL sources\n");
	/* Delete all the OpenAL sources */
	for(i=0; i<Number_of_Channels_global; i++)
	{
	fprintf(stderr, "Deleting OpenAL source: %d\n", ALmixer_Channel_List[i].alsource);
		alDeleteSources(1, &ALmixer_Channel_List[i].alsource);
	}
	/* Delete all the channels */
	free(ALmixer_Channel_List);
	free(Source_Map_List);
	
	/* Reset the Number_of_Channels just in case somebody
	 * tries using a ALmixer function.
	 * I probably should put "Initialized" checks everywhere,
	 * but I'm too lazy at the moment.
	 */
	Number_of_Channels_global = 0;
	
	context = alcGetCurrentContext();
	if(NULL == context)
	{
		return;
	}
	/* Need to get the device before I close the context */
	dev = alcGetContextsDevice(context);
	alcDestroyContext(context);
	
	if(NULL == dev)
	{
		return;	
	}
	alcCloseDevice(dev);
	
	Sound_Quit();

	return;
}

Uint8 ALmixer_IsInitialized()
{
	return ALmixer_Initialized;
}

Uint32 ALmixer_GetFrequency()
{
	return ALmixer_Frequency_global;
}

const SDL_version* ALmixer_Linked_Version()
{
	static SDL_version linked_mixver;
	ALMIXER_VERSION(&linked_mixver);
	return(&linked_mixver);
}





#if 0
void ALmixer_Output_Attributes()
{
	ALint num_flags = 0;
	ALint* flags = 0;
	int i;
	ALCdevice* dev = alcGetContextsDevice( alcGetCurrentContext() );
	

	printf("custom context\n");

	alcGetIntegerv(dev, ALC_ATTRIBUTES_SIZE,
				   sizeof num_flags, &num_flags );

	printf("Number of Flags: %d\n", num_flags);

	if(num_flags)
	{
		flags = malloc(sizeof(num_flags) * sizeof(ALint));

		alcGetIntegerv(dev, ALC_ALL_ATTRIBUTES,
		   sizeof num_flags * sizeof(ALint),
		   flags );
	}
	for(i = 0; i < num_flags-1; i += 2)
	{
		printf("key 0x%x : value %d\n",
		   flags[i], flags[i+1]);
	}
	free(flags);
}
#endif


void ALmixer_Output_Decoders()
{
	Sound_Version sound_compile_version;
	Sound_Version sound_link_version;
	
	const Sound_DecoderInfo **rc = Sound_AvailableDecoders();
	const Sound_DecoderInfo **i;
	const char **ext;
	FILE* stream = stdout;

	
	fprintf(stream, "SDL_sound Information:\n");

	SOUND_VERSION(&sound_compile_version);
	fprintf(stream, "\tCompiled with SDL_sound version: %d.%d.%d\n",
		sound_compile_version.major,
		sound_compile_version.minor,
		sound_compile_version.patch);

	Sound_GetLinkedVersion(&sound_link_version);
	fprintf(stream, "\tRunning (linked) with SDL_sound version: %d.%d.%d\n",
		sound_link_version.major,
		sound_link_version.minor,
		sound_link_version.patch);

	fprintf(stream, "Supported sound formats:\n");
	if (rc == NULL)
		fprintf(stream, " * Apparently, NONE!\n");
	else
	{
		for (i = rc; *i != NULL; i++)
		{
			fprintf(stream, " * %s\n", (*i)->description);

			for (ext = (*i)->extensions; *ext != NULL; ext++)
				fprintf(stream, "   File extension \"%s\"\n", *ext);

			fprintf(stream, "   Written by %s.\n   %s\n\n",
				(*i)->author, (*i)->url);
		} /* for */
	} /* else */

	fprintf(stream, "\n");
} 

void ALmixer_Output_OpenAL_Info()
{
	SDL_version mixer_compile_version;
	const SDL_version * mixer_link_version=ALmixer_Linked_Version();
	FILE* stream = stdout;

	fprintf(stream, "OpenAL Information:\n");
	fprintf(stream, "\tAL_VENDOR: %s\n", alGetString( AL_VENDOR ) );
	fprintf(stream, "\tAL_VERSION: %s\n", alGetString( AL_VERSION ) );
	fprintf(stream, "\tAL_RENDERER: %s\n", alGetString( AL_RENDERER ) );
	fprintf(stream, "\tAL_EXTENSIONS: %s\n", alGetString( AL_EXTENSIONS ) );

	ALMIXER_VERSION(&mixer_compile_version);
	fprintf(stream, "\nSDL_ALmixer Information:\n");
	fprintf(stream, "\tCompiled with SDL_ALmixer version: %d.%d.%d\n",
		mixer_compile_version.major,
		mixer_compile_version.minor,
		mixer_compile_version.patch);

	fprintf(stream, "\tRunning (linked) with SDL_ALmixer version: %d.%d.%d\n",
		mixer_link_version->major,
		mixer_link_version->minor,
		mixer_link_version->patch);
	
	fprintf(stream, "\tCompile flags: ");
	#ifdef ENABLE_LOKI_QUEUE_FIX_HACK
		fprintf(stream, "ENABLE_LOKI_QUEUE_FIX_HACK ");
	#endif
	#ifdef ENABLE_ALMIXER_THREADS
		fprintf(stream, "ENABLE_ALMIXER_THREADS ");
	#endif
	#ifdef ENABLE_ALC_SYNC
		fprintf(stream, "ENABLE_ALC_SYNC ");
	#endif
	fprintf(stream, "\n");
}


Sint32 ALmixer_AllocateChannels(Sint32 numchans)
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_AllocateChannels(numchans);
	SDL_UnlockMutex(simple_lock);
	return retval;
}

	
Sint32 ALmixer_ReserveChannels(Sint32 num)
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_ReserveChannels(num);
	SDL_UnlockMutex(simple_lock);
	return retval;
}

	


static ALmixer_Data* DoLoad(Sound_Sample* sample, Uint32 buffersize, Uint8 decode_mode, Uint32 max_queue_buffers, Uint32 num_startup_buffers, Uint8 access_data)
{
	Uint32 bytes_decoded;
	ALmixer_Data* ret_data;
	ALenum error;

	/* Allocate memory */
	ret_data = (ALmixer_Data *)malloc(sizeof(ALmixer_Data));
	if (NULL == ret_data) 
	{
		ALmixer_SetError("Out of memory");
		return(NULL);
	}

	/* Initialize the data fields */
	
	/* Set the Sound_Sample pointer */
	ret_data->sample = sample;

	/* Flag the data to note that it is not in use */
	ret_data->in_use = 0;

	/* Initialize remaining flags */
	ret_data->total_time = -1;
	ret_data->eof = 0;

	/* Just initialize */
	ret_data->num_buffers_in_use = 0;

	/* Just initialize */
	ret_data->total_bytes = 0;

	/* Just initialize */
	ret_data->loaded_bytes = 0;
	
	/* Set the max queue buffers (minimum must be 2) */
	if(max_queue_buffers < 2)
	{
		max_queue_buffers = ALMIXER_DEFAULT_QUEUE_BUFFERS;
	}
	ret_data->max_queue_buffers = max_queue_buffers;
	/* Set up the start up buffers */
	if(0 == num_startup_buffers)
	{
		num_startup_buffers = ALMIXER_DEFAULT_STARTUP_BUFFERS;
	}
	/* Make sure start up buffers is less or equal to max_queue_buffers */
	if(num_startup_buffers > max_queue_buffers)
	{
		num_startup_buffers = max_queue_buffers;
	}
	ret_data->num_startup_buffers = num_startup_buffers;
	
	ret_data->buffer_map_list = NULL;
	ret_data->current_buffer = 0;
	
	ret_data->circular_buffer_queue = NULL;
	
	/* Now decode and load the data into a data chunk */
	/* Different cases for Streamed and Predecoded 
	 * Streamed might turn into a predecoded if buffersize
	 * is large enough */
	if(ALMIXER_DECODE_STREAM == decode_mode)
	{
		bytes_decoded = Sound_Decode(sample);
		if(sample->flags & SOUND_SAMPLEFLAG_ERROR)
		{
			ALmixer_SetError(Sound_GetError());
			Sound_FreeSample(sample);
			free(ret_data);
			return NULL;
		}

		/* If no data, return an error */
		if(0 == bytes_decoded)
		{
			ALmixer_SetError("File has no data");
			Sound_FreeSample(sample);
			free(ret_data);
			return NULL;
		}
		
		/* Note, currently, my Ogg conservative modifications
		 * prevent EOF from being detected in the first read
		 * because of the weird packet behavior of ov_read().
		 * The EAGAIN will get set, but not the EOF.
		 * I don't know the best way to handle this,
		 * so for now, Ogg's can only be explicitly
		 * predecoded.
		 */

		/* Correction: Since we no longer actually keep the 
		 * streamed data we read here (we rewind and throw
		 * it away, and start over on Play), it is
		 * safe to read another chunk to see if we've hit EOF
		 */
		if(sample->flags & SOUND_SAMPLEFLAG_EAGAIN)
		{
			bytes_decoded = Sound_Decode(sample);
			if(sample->flags & SOUND_SAMPLEFLAG_ERROR)
			{
				ALmixer_SetError(Sound_GetError());
				Sound_FreeSample(sample);
				free(ret_data);
				return NULL;
			}
		}


		/* If we found an EOF, the entire file was 
		 * decoded, so we can treat it like one.
		 */
		
		if(sample->flags & SOUND_SAMPLEFLAG_EOF)
		{
	fprintf(stderr, "We got LUCKY! File is predecoded even though STREAM was requested\n");

			ret_data->decoded_all = 1;
			/* Need to keep this information around for
			 * seek and rewind abilities.
			 */
			ret_data->total_bytes = bytes_decoded;
			/* For now, the loaded bytes is the same as total bytes, but
			 * this could change during a seek operation
			 */
			ret_data->loaded_bytes = bytes_decoded;

			/* Let's compute the total playing time 
			 * SDL_sound does not yet provide this (we're working on
			 * that at the moment...)
			 */
			ret_data->total_time = Compute_Total_Time(&sample->desired, bytes_decoded);

			/* Create one element in the buffer array for data for OpanAL */
			ret_data->buffer = (ALuint*)malloc( sizeof(ALuint) );
			if(NULL == ret_data->buffer)
			{
				ALmixer_SetError("Out of Memory");
				Sound_FreeSample(sample);
				free(ret_data);
				return NULL;
			}
			/* Clear the error code */
			alGetError();
			/* Now generate an OpenAL buffer using that first element */
			alGenBuffers(1, ret_data->buffer);
			if( (error = alGetError()) != AL_NO_ERROR)
			{
				ALmixer_SetError("alGenBuffers failed: %s\n", aluGetErrorString(error));
				Sound_FreeSample(sample);
				free(ret_data->buffer);
				free(ret_data);
				return NULL;
			}
			
			
			/* Now copy the data to the OpenAL buffer */
			/* We can't just set a pointer because the API needs
			 * its own copy to assist hardware acceleration */
			alBufferData(ret_data->buffer[0], 
				TranslateFormat(&sample->desired), 
				sample->buffer,
				bytes_decoded,
				sample->desired.rate
			);
			if( (error = alGetError()) != AL_NO_ERROR)
			{
				ALmixer_SetError("alBufferData failed: %s\n", aluGetErrorString(error));
				Sound_FreeSample(sample);
				alDeleteBuffers(1, ret_data->buffer);
				free(ret_data->buffer);
				free(ret_data);
				return NULL;
			}

			/* We should be done with the sample since it's all 
			 * predecoded. So we can free the memory */
			
			/* Additional notes:
			 * We need to keep data around in case Seek() is needed
			 * or other Sound_AudioInfo is needed.
			 * This can either be done by not deleting the sample,
			 * or it can be done by dynamically recreating it
			 * when we need it. 
			 */
			/* Since OpenAL won't let us retrieve it
			 * (aka dynamically), we have to keep the Sample
			 * around because since the user requested
			 * streamed and we offered predecoded,
			 * we don't want to mess up the user who
			 * was expecting seek support
			 * So Don't Do anything
			 */
			/*
			if(0 == access_data)
			{
				Sound_FreeSample(sample);
				ret_data->sample = NULL;
			}
			*/	
			/* Else, We keep a copy of the sample around.
			 * so don't do anything.
			 */
					
#if 0
#if defined(DISABLE_PREDECODED_SEEK)
			Sound_FreeSample(sample);
			ret_data->sample = NULL;
#elif !defined(DISABLE_SEEK_MEMORY_OPTIMIZATION)
			Sound_FreeSample(sample);
			ret_data->sample = NULL;
#else
			/* We keep a copy of the sample around.
			 * so don't do anything.
			 */
#endif
#endif
			/* okay we're done here */
			
		}
		/* Else, we need to stream the data, so we'll
		 * create multple buffers for queuing */
		else
		{
	fprintf(stderr, "Loading streamed data (not lucky)\n");
			ret_data->decoded_all = 0;

			/* This information is for predecoded.
			 * Set to 0, since we don't know.
			 */
			ret_data->total_bytes = 0;

			/* Create buffers for data
			 */
			ret_data->buffer = (ALuint*)malloc( sizeof(ALuint) * max_queue_buffers);
			if(NULL == ret_data->buffer)
			{
				ALmixer_SetError("Out of Memory");
				Sound_FreeSample(sample);
				free(ret_data);
				return NULL;
			}

			/* Clear the error code */
			alGetError();
			/* Now generate an OpenAL buffer using that first element */
			alGenBuffers(max_queue_buffers, ret_data->buffer);
			if( (error = alGetError()) != AL_NO_ERROR)
			{
				ALmixer_SetError("alGenBuffers failed: %s\n", aluGetErrorString(error));
				Sound_FreeSample(sample);
				free(ret_data->buffer);
				free(ret_data);
				return NULL;
			}
			
			/* Redesign: Okay, because of the unqueuing problems and such,
			 * I've decided to redesign where and how queuing is handled.
			 * Before, everything was queued up here. However, this
			 * placed a penalty on load and made performance inconsistent
			 * when samples had to be rewound. It did make things easier
			 * to queue because I could let OpenAL decide which buffer
			 * needed to be queued next.
			 * Now, I'm going to push off the queuing to the actual
			 * Play() command. I'm going to add some book keeping,
			 * and allow for additional buffers to be filled at later
			 * times. 
			 */


			/* So first of all, because of I already decoded the sample
			 * for testing, I need to decide what to do with it.
			 * The best thing would be be to alBufferData() it.
			 * The problem is it may conflict with the rest of 
			 * the system because everything now assumes buffers
			 * are entirely stripped (because of the unqueing
			 * problem).
			 * So it looks like I have to do the crappy thing 
			 * and throw away the data, and rewind.
			 */
			
			if(0 == Sound_Rewind(ret_data->sample))
			{
				ALmixer_SetError("Cannot use sample for streamed data because it must be rewindable: %s", Sound_GetError() );
				Sound_FreeSample(sample);
				free(ret_data->buffer);
				free(ret_data);
				return NULL;
			}
			

			/* If the user has selected access_data, we need to 
			 * keep copies of the queuing buffers around because
			 * OpenAL won't let us access the data.
			 * Allocate the memory for the buffers here
			 * and initialize the albuffer-index map
			 */
			if(access_data)
			{
				Uint32 j;
				/* Create buffers for data access
				 * Should be the same number as the number of queue buffers
				 */
				ret_data->buffer_map_list = (Buffer_Map*)malloc( sizeof(Buffer_Map) * max_queue_buffers);
				if(NULL == ret_data->buffer_map_list)
				{
					ALmixer_SetError("Out of Memory");
					Sound_FreeSample(sample);
					free(ret_data->buffer);
					free(ret_data);
					return NULL;
				}

				ret_data->circular_buffer_queue = CircularQueueUnsignedInt_CreateQueue(max_queue_buffers);
				if(NULL == ret_data->circular_buffer_queue)
				{
					ALmixer_SetError("Out of Memory");
					free(ret_data->buffer_map_list);
					Sound_FreeSample(sample);
					free(ret_data->buffer);
					free(ret_data);
					return NULL;
				}


				for(j=0; j<max_queue_buffers; j++)
				{
					ret_data->buffer_map_list[j].albuffer = ret_data->buffer[j];
					ret_data->buffer_map_list[j].index = j;
					ret_data->buffer_map_list[j].num_bytes = 0;
					ret_data->buffer_map_list[j].data = (Uint8*)malloc( sizeof(Uint8) * buffersize);
					if(NULL == ret_data->buffer_map_list[j].data)
					{
						ALmixer_SetError("Out of Memory");
						break;
					}
				}
				/* If an error happened, we have to clean up the memory */
				if(j < max_queue_buffers)
				{
					fprintf(stderr, "################## Buffer allocation failed\n");
					for( ; j>=0; j--)
					{
						free(ret_data->buffer_map_list[j].data);
					}
					free(ret_data->buffer_map_list);
					CircularQueueUnsignedInt_FreeQueue(ret_data->circular_buffer_queue);
					Sound_FreeSample(sample);
					free(ret_data->buffer);
					free(ret_data);
					return NULL;
				}

				/* The Buffer_Map_List must be sorted by albuffer for binary searches
	 			*/
				qsort(ret_data->buffer_map_list, max_queue_buffers, sizeof(Buffer_Map), Compare_Buffer_Map);
			} /* End if access_data==true */

			
		} /* End of do stream */
	} /* end of DECODE_STREAM */
	/* User requested decode all (easy, nothing to figure out) */
	else if(ALMIXER_DECODE_ALL == decode_mode)
	{
		bytes_decoded = Sound_DecodeAll(sample);
		if(sample->flags & SOUND_SAMPLEFLAG_ERROR)
		{
			ALmixer_SetError(Sound_GetError());
			Sound_FreeSample(sample);
			free(ret_data);
			return NULL;
		}

		/* If no data, return an error */
		if(0 == bytes_decoded)
		{
			ALmixer_SetError("File has no data");
			Sound_FreeSample(sample);
			free(ret_data);
			return NULL;
		}
		

		ret_data->decoded_all = 1;
		/* Need to keep this information around for
		 * seek and rewind abilities.
		 */
		ret_data->total_bytes = bytes_decoded;
		/* For now, the loaded bytes is the same as total bytes, but
		 * this could change during a seek operation
		 */
		ret_data->loaded_bytes = bytes_decoded;

		/* Let's compute the total playing time 
		 * SDL_sound does not yet provide this (we're working on
		 * that at the moment...)
		 */
		ret_data->total_time = Compute_Total_Time(&sample->desired, bytes_decoded);

		/* Create one element in the buffer array for data for OpanAL */
		ret_data->buffer = (ALuint*)malloc( sizeof(ALuint) );
		if(NULL == ret_data->buffer)
		{
			ALmixer_SetError("Out of Memory");
			Sound_FreeSample(sample);
			free(ret_data);
			return NULL;
		}
		/* Clear the error code */
		alGetError();
		/* Now generate an OpenAL buffer using that first element */
		alGenBuffers(1, ret_data->buffer);
		if( (error = alGetError()) != AL_NO_ERROR)
		{
			ALmixer_SetError("alGenBuffers failed: %s\n", aluGetErrorString(error));
			Sound_FreeSample(sample);
			free(ret_data->buffer);
			free(ret_data);
			return NULL;
		}
		fprintf(stderr, "Actual rate=%d, desired=%d\n", sample->actual.rate, sample->desired.rate);

		/* Now copy the data to the OpenAL buffer */
		/* We can't just set a pointer because the API needs
		 * its own copy to assist hardware acceleration */
		alBufferData(ret_data->buffer[0], 
			TranslateFormat(&sample->desired), 
			sample->buffer,
			bytes_decoded,
			sample->desired.rate
		);
		if( (error = alGetError()) != AL_NO_ERROR)
		{
			ALmixer_SetError("alBufferData failed: %s\n", aluGetErrorString(error));
			Sound_FreeSample(sample);
			alDeleteBuffers(1, ret_data->buffer);
			free(ret_data->buffer);
			free(ret_data);
			return NULL;
		}

		/* We should be done with the sample since it's all 
		 * predecoded. So we can free the memory */
		/* Need to keep around because Seek() needs it */
		
		/* Additional notes:
		 * We need to keep data around in case Seek() is needed
		 * or other Sound_AudioInfo is needed.
		 * This can either be done by not deleting the sample,
		 * or it can be done by dynamically recreating it
		 * when we need it. 
		 * Update: I think now it's up to the user by passing the 
		 * access_data flag. If they set the flag, then they get 
		 * data callbacks and seek support. If not, then they can
		 * get all that stuff at the expense of keeping extra memory 
		 * around.
		 */
		if(0 == access_data)
		{
			Sound_FreeSample(sample);
			ret_data->sample = NULL;
		}
			
		/* Else, We keep a copy of the sample around.
		 * so don't do anything.
		 */
#if 0
#if defined(DISABLE_PREDECODED_SEEK) 
		Sound_FreeSample(sample);
		ret_data->sample = NULL;
#elif !defined(DISABLE_SEEK_MEMORY_OPTIMIZATION)
		Sound_FreeSample(sample);
		ret_data->sample = NULL;
#else
		/* We keep a copy of the sample around.
		 * so don't do anything.
		 */
#endif
#endif

fprintf(stderr, "Made it\n");
		/* okay we're done here */
	}
	else
	{
		/* Shouldn't get here */
		ALmixer_SetError("Unknown decode mode");
		Sound_FreeSample(sample);
		free(ret_data);
		return NULL;
	}
		
fprintf(stderr, "Returning data\n");
	return ret_data;
}


/* This will load a sample for us. Most of the uglyness is
 * error checking and the fact that streamed/predecoded files
 * must be treated differently.
 * I don't like the AudioInfo parameter. I removed it once,
 * but the system will fail on RAW samples because the user
 * must specify it, so I had to bring it back.
 * Remember I must close the rwops if there is an error before NewSample()
 */
ALmixer_Data* ALmixer_LoadSample_RW(SDL_RWops* rwops, const char* fileext, Uint32 buffersize, Uint8 decode_mode, Uint32 max_queue_buffers, Uint32 num_startup_buffers, Uint8 access_data)
{
	Sound_Sample* sample = NULL;
	Sound_AudioInfo target;
	
	/* Initialize target values to defaults 
	 * 0 tells SDL_sound to use the "actual" values
	 */
	target.channels = 0;
	target.rate = 0;
#if 0
	/* This requires my new additions to SDL_sound. It will
	 * convert the sample to the proper endian order.
	 * If the actual is 8-bit, it will do unsigned, if 
	 * the actual is 16-bit, it will do signed.
	 * I'm told by Ryan Gordon that OpenAL prefers the signedness
	 * in this way.
	 */
	target.format = AUDIO_U8S16SYS;
#else
	target.format = AUDIO_S16SYS;
#endif
	
	/* Set a default buffersize if needed */
	if(0 == buffersize)
	{
		buffersize = ALMIXER_DEFAULT_BUFFERSIZE;
	}
	
	sample = Sound_NewSample(rwops, fileext, &target, buffersize);
	if(NULL == sample)
	{
		ALmixer_SetError(Sound_GetError());
		return NULL;
	}

	return( DoLoad(sample, buffersize, decode_mode, max_queue_buffers, num_startup_buffers, access_data));
}



/* This will load a sample for us from 
 * a file (instead of RWops). Most of the uglyness is
 * error checking and the fact that streamed/predecoded files
 * must be treated differently.
 */
ALmixer_Data* ALmixer_LoadSample(const char* filename, Uint32 buffersize, Uint8 decode_mode, Uint32 max_queue_buffers, Uint32 num_startup_buffers, Uint8 access_data)
{
	Sound_Sample* sample = NULL;
	Sound_AudioInfo target;
	
	/* Initialize target values to defaults 
	 * 0 tells SDL_sound to use the "actual" values
	 */
	target.channels = 0;
	target.rate = 0;
	
#if 0
	/* This requires my new additions to SDL_sound. It will
	 * convert the sample to the proper endian order.
	 * If the actual is 8-bit, it will do unsigned, if 
	 * the actual is 16-bit, it will do signed.
	 * I'm told by Ryan Gordon that OpenAL prefers the signedness
	 * in this way.
	 */
	target.format = AUDIO_U8S16SYS;
#else
	target.format = AUDIO_S16SYS;
#endif
	
#if 0	
	/* Okay, here's a messy hack. The problem is that we need
	 * to convert the sample to have the correct bitdepth,
	 * endian order, and signedness values. 
	 * The bit depth is 8 or 16.
	 * The endian order is the native order of the system.
	 * The signedness depends on what the original value
	 * of the sample. Unfortunately, we can't specify these
	 * values until we after we already know what the original
	 * values were for bitdepth and signedness. 
	 * So we must open the file once to get the values, 
	 * then close it, and then reopen it with the 
	 * correct desired target values.
	 * I tried changing the sample->desired field after 
	 * the NewSample call, but it had no effect, so 
	 * it looks like it must be set on open.
	 */
	/* Pick a small buffersize for the first open to not
	 * waste much time allocating memory */
	sample = Sound_NewSampleFromFile(filename, NULL, 512);
	if(NULL == sample)
	{
		ALmixer_SetError(Sound_GetError());
		return NULL;
	}

	bit_depth = GetBitDepth(sample->actual.format);
	signedness_value = GetSignednessValue(sample->actual.format);
	if(8 == bit_depth)
	{
		/* If 8 bit, then we don't have to worry about 
		 * endian issues. We can just use the actual format
		 * value and it should do the right thing
		 */
		target.format = sample->actual.format;
	}
	else
	{
		/* We'll assume it's 16-bit, and if it's not
		 * hopefully SDL_sound will return an error, 
		 * or let us convert to 16-bit
		 */
		/* Now we need to get the correct signedness */
		if(ALMIXER_UNSIGNED_VALUE == signedness_value)
		{
			/* Set to Unsigned 16-bit, system endian order */
			target.format = AUDIO_U16SYS;
		}
		else
		{
			/* Again, we'll assume it's Signed 16-bit system order
			 * or force the conversion and hope it works out
			 */
			target.format = AUDIO_S16SYS;
		}
	}
	
	/* Now we have the correct info. We need to close and reopen */
	Sound_FreeSample(sample);
#endif

	sample = Sound_NewSampleFromFile(filename, &target, buffersize);
	if(NULL == sample)
	{
		ALmixer_SetError(Sound_GetError());
		return NULL;
	}

		fprintf(stderr, "Correction test: Actual rate=%d, desired=%d, actual format=%d, desired format=%d\n", sample->actual.rate, sample->desired.rate, sample->actual.format, sample->desired.format);

	return( DoLoad(sample, buffersize, decode_mode, max_queue_buffers, num_startup_buffers, access_data));
}


/* This is a back door for RAW samples or if you need the
 * AudioInfo field. Use at your own risk.
 */
ALmixer_Data* ALmixer_LoadSample_RAW_RW(SDL_RWops* rwops, const char* fileext, Sound_AudioInfo* desired, Uint32 buffersize, Uint8 decode_mode, Uint32 max_queue_buffers, Uint32 num_startup_buffers, Uint8 access_data)
{
	Sound_Sample* sample = NULL;
	sample = Sound_NewSample(rwops, fileext, desired, buffersize);
	if(NULL == sample)
	{
		ALmixer_SetError(Sound_GetError());
		return NULL;
	}
	return( DoLoad(sample, buffersize, decode_mode, max_queue_buffers, num_startup_buffers, access_data));
}




/* This is a back door for RAW samples or if you need the
 * AudioInfo field. Use at your own risk.
 */
ALmixer_Data* ALmixer_LoadSample_RAW(const char* filename, Sound_AudioInfo* desired, Uint32 buffersize, Uint8 decode_mode, Uint32 max_queue_buffers, Uint32 num_startup_buffers, Uint8 access_data)
{
	Sound_Sample* sample = NULL;
	sample = Sound_NewSampleFromFile(filename, desired, buffersize);
	if(NULL == sample)
	{
		ALmixer_SetError(Sound_GetError());
		return NULL;
	}
	return( DoLoad(sample, buffersize, decode_mode, max_queue_buffers, num_startup_buffers, access_data));
}




void ALmixer_FreeData(ALmixer_Data* data)
{
	ALenum error;
	if(NULL == data)
	{
		return;
	}
	
	if(data->decoded_all)
	{
		/* If access_data was enabled, then the Sound_Sample*
		 * still exists. We need to free it
		 */
		if(data->sample != NULL)
		{
			Sound_FreeSample(data->sample);
		}
		alDeleteBuffers(1, data->buffer);
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "70Testing error: %s\n",
			aluGetErrorString(error));				
	}

	}
	else
	{
		Uint32 i;
		
		/* Delete buffer copies if access_data was enabled */
		if(data->buffer_map_list != NULL)
		{
			for(i=0; i<data->max_queue_buffers; i++)
			{
				free(data->buffer_map_list[i].data);
			}
			free(data->buffer_map_list);
		}
		if(data->circular_buffer_queue != NULL)
		{
			CircularQueueUnsignedInt_FreeQueue(data->circular_buffer_queue);
		}
			
		Sound_FreeSample(data->sample);
		alDeleteBuffers(data->max_queue_buffers, data->buffer);		
	if((error = alGetError()) != AL_NO_ERROR)
	{
		fprintf(stderr, "71Testing error: %s\n",
			aluGetErrorString(error));				
	}
	}
	free(data->buffer);
	free(data);
}

Sint32 ALmixer_GetTotalTime(ALmixer_Data* data)
{
	return data->total_time;
}

/* This function will look up the source for the corresponding channel */
/* Must return 0 on error instead of -1 because of unsigned int */
ALuint ALmixer_GetSource(Sint32 channel)
{
	ALuint retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_GetSource(channel);
	SDL_UnlockMutex(simple_lock);
	return retval;
}

/* This function will look up the channel for the corresponding source */
Sint32 ALmixer_GetChannel(ALuint source)
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_GetChannel(source);
	SDL_UnlockMutex(simple_lock);
	return retval;
}

Sint32 ALmixer_FindFreeChannel(Sint32 start_channel)
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_FindFreeChannel(start_channel);
	SDL_UnlockMutex(simple_lock);
	return retval;
}



/* API update function.
 * It should return the number of buffers that were 
 * queued during the call. The value might be
 * used to guage how long you might wait to
 * call the next update loop in case you are worried
 * about preserving CPU cycles. The idea is that
 * when a buffer is queued, there was probably some
 * CPU intensive looping which took awhile.
 * It's mainly provided as a convenience.
 * Timing the call with SDL_GetTicks() would produce
 * more accurate information.
 * Returns a negative value if there was an error,
 * the value being the number of errors.
 */
Sint32 ALmixer_Update()
{
#ifdef ENABLE_ALMIXER_THREADS
	/* The thread will handle all updates by itself.
	 * Don't allow the user to explicitly call update.
	 */
	return 0;
#else
	return( Update_ALmixer(NULL) );
#endif
}



void ALmixer_ChannelFinished(void (*channel_finished)(Sint32 channel, void* userdata), void* userdata)
{
	SDL_LockMutex(simple_lock);
	Channel_Done_Callback = channel_finished;
	Channel_Done_Callback_Userdata = userdata;
	SDL_UnlockMutex(simple_lock);
}


void ALmixer_ChannelData(void (*channel_data)(Sint32 which_chan, Uint8* data, Uint32 num_bytes, Uint32 frequency, Uint8 channels, Uint8 bitdepth, Uint16 format, Uint8 decode_mode))
{
	SDL_LockMutex(simple_lock);
	Channel_Data_Callback = channel_data;
	SDL_UnlockMutex(simple_lock);
}





Sint32 ALmixer_PlayChannelTimed(Sint32 channel, ALmixer_Data* data, Sint32 loops, Sint32 ticks)
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_PlayChannelTimed(channel, data, loops, ticks);
	SDL_UnlockMutex(simple_lock);
	return retval;
}


/* In case the user wants to specify a source instead of a channel,
 * they may use this function. This function will look up the 
 * source-to-channel map, and convert the call into a
 * PlayChannelTimed() function call.
 * Returns the channel it's being played on.
 * Note: If you are prefer this method, then you need to be careful
 * about using PlayChannel, particularly if you request the
 * first available channels because source and channels have 
 * a one-to-one mapping in this API. It is quite easy for 
 * a channel/source to already be in use because of this.
 * In this event, an error message will be returned to you.
 */
ALuint ALmixer_PlaySourceTimed(ALuint source, ALmixer_Data* data, Sint32 loops, Sint32 ticks)
{
	ALuint retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_PlaySourceTimed(source, data, loops, ticks);
	SDL_UnlockMutex(simple_lock);
	return retval;
}


/* Will return the number of channels halted
 * or 0 for error
 */
Sint32 ALmixer_HaltChannel(Sint32 channel)
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_HaltChannel(channel);
	SDL_UnlockMutex(simple_lock);
	return retval;
}

/* Will return the number of channels halted
 * or 0 for error
 */
Sint32 ALmixer_HaltSource(ALuint source)
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_HaltSource(source);
	SDL_UnlockMutex(simple_lock);
	return retval;
}


/* This will rewind the SDL_Sound sample for streamed
 * samples and start buffering up the data for the next
 * playback. This may require samples to be halted
 */
Sint32 ALmixer_RewindData(ALmixer_Data* data)
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_RewindData(data);
	SDL_UnlockMutex(simple_lock);
	return retval;
}

Sint32 ALmixer_RewindChannel(Sint32 channel)
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_RewindChannel(channel);
	SDL_UnlockMutex(simple_lock);
	return retval;
}

Sint32 ALmixer_RewindSource(ALuint source)
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_RewindSource(source);
	SDL_UnlockMutex(simple_lock);
	return retval;
}

Sint32 ALmixer_PauseChannel(Sint32 channel)
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_PauseChannel(channel);
	SDL_UnlockMutex(simple_lock);
	return retval;
}

Sint32 ALmixer_PauseSource(ALuint source)
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_PauseSource(source);
	SDL_UnlockMutex(simple_lock);
	return retval;
}

Sint32 ALmixer_ResumeChannel(Sint32 channel)
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_ResumeChannel(channel);
	SDL_UnlockMutex(simple_lock);
	return retval;
}

Sint32 ALmixer_ResumeSource(ALuint source)
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_ResumeSource(source);
	SDL_UnlockMutex(simple_lock);
	return retval;
}

/* Might consider setting eof to 0 as a "feature"
 * This will allow seek to end to stay there because
 * Play automatically rewinds if at the end */
Sint32 ALmixer_Seek(ALmixer_Data* data, Uint32 msec)
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_Seek(data, msec);
	SDL_UnlockMutex(simple_lock);
	return retval;
}

Sint32 ALmixer_FadeInChannelTimed(Sint32 channel, ALmixer_Data* data, Sint32 loops, Uint32 fade_ticks, Sint32 expire_ticks)
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_FadeInChannelTimed(channel, data, loops, fade_ticks, expire_ticks);
	SDL_UnlockMutex(simple_lock);
	return retval;
}

ALuint ALmixer_FadeInSourceTimed(ALuint source, ALmixer_Data* data, Sint32 loops, Uint32 fade_ticks, Sint32 expire_ticks)
{
	ALuint retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_FadeInSourceTimed(source, data, loops, fade_ticks, expire_ticks);
	SDL_UnlockMutex(simple_lock);
	return retval;
}

Sint32 ALmixer_FadeOutChannel(Sint32 channel, Uint32 ticks)
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_FadeOutChannel(channel, ticks);
	SDL_UnlockMutex(simple_lock);
	return retval;
}
	
Sint32 ALmixer_FadeOutSource(ALuint source, Uint32 ticks)
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_FadeOutSource(source, ticks);
	SDL_UnlockMutex(simple_lock);
	return retval;
}

Sint32 ALmixer_FadeChannel(Sint32 channel, Uint32 ticks, ALfloat volume)
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_FadeChannel(channel, ticks, volume);
	SDL_UnlockMutex(simple_lock);
	return retval;
}

Sint32 ALmixer_FadeSource(ALuint source, Uint32 ticks, ALfloat volume)
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_FadeSource(source, ticks, volume);
	SDL_UnlockMutex(simple_lock);
	return retval;
}

Sint32 ALmixer_SetMaxVolumeChannel(Sint32 channel, ALfloat volume)
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_SetMaxVolumeChannel(channel, volume);
	SDL_UnlockMutex(simple_lock);
	return retval;
}

Sint32 ALmixer_SetMaxVolumeSource(ALuint source, ALfloat volume)
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_SetMaxVolumeSource(source, volume);
	SDL_UnlockMutex(simple_lock);
	return retval;
}

ALfloat ALmixer_GetMaxVolumeChannel(Sint32 channel)
{
	ALfloat retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_GetMaxVolumeChannel(channel);
	SDL_UnlockMutex(simple_lock);
	return retval;	
}

ALfloat ALmixer_GetMaxVolumeSource(ALuint source)
{
	ALfloat retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_GetMaxVolumeSource(source);
	SDL_UnlockMutex(simple_lock);
	return retval;	
}


Sint32 ALmixer_SetMinVolumeChannel(Sint32 channel, ALfloat volume)
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_SetMinVolumeChannel(channel, volume);
	SDL_UnlockMutex(simple_lock);
	return retval;
}

Sint32 ALmixer_SetMinVolumeSource(ALuint source, ALfloat volume)
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_SetMinVolumeSource(source, volume);
	SDL_UnlockMutex(simple_lock);
	return retval;
}

ALfloat ALmixer_GetMinVolumeChannel(Sint32 channel)
{
	ALfloat retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_GetMinVolumeChannel(channel);
	SDL_UnlockMutex(simple_lock);
	return retval;	
}

ALfloat ALmixer_GetMinVolumeSource(ALuint source)
{
	ALfloat retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_GetMinVolumeSource(source);
	SDL_UnlockMutex(simple_lock);
	return retval;	
}



Sint32 ALmixer_SetMasterVolume(ALfloat volume)
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_SetMasterVolume(volume);
	SDL_UnlockMutex(simple_lock);
	return retval;	
}

ALfloat ALmixer_GetMasterVolume()
{
	ALfloat retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_GetMasterVolume();
	SDL_UnlockMutex(simple_lock);
	return retval;	
}

Sint32 ALmixer_ExpireChannel(Sint32 channel, Sint32 ticks)
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_ExpireChannel(channel, ticks);
	SDL_UnlockMutex(simple_lock);
	return retval;
}

Sint32 ALmixer_ExpireSource(ALuint source, Sint32 ticks)
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_ExpireSource(source, ticks);
	SDL_UnlockMutex(simple_lock);
	return retval;
}

Sint32 ALmixer_QueryChannel(Sint32 channel)
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_QueryChannel(channel);
	SDL_UnlockMutex(simple_lock);
	return retval;
}

Sint32 ALmixer_QuerySource(ALuint source)
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_QuerySource(source);
	SDL_UnlockMutex(simple_lock);
	return retval;
}


Sint32 ALmixer_PlayingChannel(Sint32 channel)
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_PlayingChannel(channel);
	SDL_UnlockMutex(simple_lock);
	return retval;
}

Sint32 ALmixer_PlayingSource(ALuint source)
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_PlayingSource(source);
	SDL_UnlockMutex(simple_lock);
	return retval;
}


Sint32 ALmixer_PausedChannel(Sint32 channel)
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_PausedChannel(channel);
	SDL_UnlockMutex(simple_lock);
	return retval;
}

Sint32 ALmixer_PausedSource(ALuint source)
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_PausedSource(source);
	SDL_UnlockMutex(simple_lock);
	return retval;
}


Sint32 ALmixer_CountAllFreeChannels()
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_CountAllFreeChannels();
	SDL_UnlockMutex(simple_lock);
	return retval;
}

Sint32 ALmixer_CountUnreservedFreeChannels()
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_CountUnreservedFreeChannels();
	SDL_UnlockMutex(simple_lock);
	return retval;
}

Sint32 ALmixer_CountAllUsedChannels()
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_CountAllUsedChannels();
	SDL_UnlockMutex(simple_lock);
	return retval;
}

Sint32 ALmixer_CountUnreservedUsedChannels()
{
	Sint32 retval;
	SDL_LockMutex(simple_lock);
	retval = Internal_CountUnreservedUsedChannels();
	SDL_UnlockMutex(simple_lock);
	return retval;
}