almixer_isolated: SDL_ALmixer.c comparison

comparison SDL_ALmixer.c @ 0:01e39f9f58d5

Bitkeeper era.

author	Eric Wing <ewing . public \|-at-\| gmail . com>
date	Wed, 27 Oct 2010 16:50:19 -0700
parents
children	a8a8fe374984

comparison

equal deleted inserted replaced

--1:000000000000
+:01e39f9f58d5
+/* 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;
+}

Mercurial > almixer_isolated

comparison SDL_ALmixer.c @ 0:01e39f9f58d5