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	diff --git a/util/audio.cpp b/util/audio.cpp
	index 405cda7e..0ddebb8a 100644
	--- a/util/audio.cpp
	+++ b/util/audio.cpp
	@@ -1,191 +1,261 @@
	#include <string.h>

	#include "audio.h"
	#include "debug.h"

	namespace Util{

	#ifdef USE_SDL1
	AudioConverter::AudioConverter(Encoding inputEncoding, int inputChannels, int inputFrequency,
	Encoding outputEncoding, int outputChannels, int outputFrequency){

	SDL_BuildAudioCVT(&conversion, inputEncoding, inputChannels, inputFrequency,
	outputEncoding, outputChannels, outputFrequency);
	}

	AudioConverter::~AudioConverter(){
	}

	int AudioConverter::convertedLength(int length){
	return length;
	}

	int AudioConverter::convert(void * input, int length){
	if (conversion.needed){
	conversion.buf = (Uint8*) input;
	conversion.len = length;
	/* then convert to whatever the real output wants */
	SDL_ConvertAudio(&conversion);
	return conversion.len_cvt;
	} else {
	return length;
	}
	}
	#else
	AudioConverter::AudioConverter(Encoding inputEncoding, int inputChannels, int inputFrequency,
	Encoding outputEncoding, int outputChannels, int outputFrequency):
	buffer(NULL),
	bufferSize(0){

	input.bytes = inputEncoding;
	input.channels = inputChannels;
	input.frequency = inputFrequency;

	output.bytes = outputEncoding;
	output.channels = outputChannels;
	output.frequency = outputFrequency;

	sizeRatio = (double) byteSize(output) * output.frequency / ((double) byteSize(input) * input.frequency);
	}

	int AudioConverter::byteSize(const Format & what){
	return encodingBytes(what.bytes) * what.channels;
	}

	/* how many bytes an encoding takes up */
	int AudioConverter::encodingBytes(Encoding what){
	switch (what){
	+ case Unsigned8: return 1;
	case Signed16: return 2;
	+ case Unsigned16: return 2;
	case Float32: return 4;
	}

	return 1;
	}

	int AudioConverter::convertedLength(int length){
	// return length * sizeRatio;

	int total = length * sizeRatio;

	/* make sure we get an even number of samples */
	if (total % byteSize(output) != 0){
	total -= total % byteSize(output);
	}
	return total;
	}

	-double CubicInterpolate(double y0,double y1,
	- double y2,double y3,
	+double CubicInterpolate(double y0, double y1,
	+ double y2, double y3,
	double mu){
	double a0,a1,a2,a3,mu2;

	mu2 = mu*mu;
	a0 = y3 - y2 - y0 + y1;
	a1 = y0 - y1 - a0;
	a2 = y2 - y0;
	a3 = y1;

	return (a0mumu2+a1mu2+a2mu+a3);
	}

	-template <class Size>
	-Size clamp(double input){
	- Size top = (1 << (sizeof(Size) * 8 - 1)) - 1;
	- Size bottom = -(1 << (sizeof(Size) * 8 - 1));
	+template <class Input, class Output>
	+Output clamp(double input){
	+ Output top = (1 << (sizeof(Output) * 8 - 1)) - 1;
	+ Output bottom = -(1 << (sizeof(Output) * 8 - 1));
	if (input > top){
	return top;
	}
	if (input < bottom){
	return bottom;
	}
	return input;
	}

	template <>
	-float clamp(double input){
	+float clamp<float>(double input){
	return input;
	}

	-template <class Size>
	-void doConvertRate(Size * input, int inputSamples, int inputChannels, Size * buffer, int outputSamples, int outputChannels, double ratio){
	+template <>
	+unsigned short clamp<signed short>(double input){
	+ return (int) clamp<signed short, signed short>(input) + (int) (1 << (sizeof(signed short) * 8 - 1));
	+}
	+
	+template <>
	+unsigned char clamp<signed short>(double input){
	+ double out = input / (1 << (sizeof(signed short) * 8 - 1));
	+ if (out > 1){
	+ return 1;
	+ }
	+ if (out < -1){
	+ return -1;
	+ }
	+ /* -1,1 -> 0,255 */
	+ return (unsigned char)((out + 1) * 128);
	+}
	+
	+template <>
	+float clamp<signed short>(double input){
	+ double out = input / (1 << (sizeof(signed short) * 8 - 1));
	+ if (out > 1){
	+ return 1;
	+ }
	+ if (out < -1){
	+ return -1;
	+ }
	+ return out;
	+}
	+
	+template <class SizeInput, class SizeOutput>
	+void doConvertRate(SizeInput * input, int inputSamples, int inputChannels, SizeOutput * buffer, int outputSamples, int outputChannels, double ratio){
	for (int sample = 0; sample < outputSamples; sample += 1){
	double inputSample = sample / ratio;

	for (int channel = 0; channel < outputChannels; channel += 1){

	int inputChannel = inputChannels > channel ? channel : inputChannels - 1;

	int sample0 = ((int) inputSample - 1) * inputChannels + inputChannel;
	int sample1 = ((int) inputSample + 0) * inputChannels + inputChannel;
	int sample2 = ((int) inputSample + 1) * inputChannels + inputChannel;
	int sample3 = ((int) inputSample + 2) * inputChannels + inputChannel;

	if (sample0 < 0){
	sample0 = sample1;
	}

	if (sample2 >= inputSamples * inputChannels){
	sample2 = sample1;
	}

	if (sample3 >= inputSamples * inputChannels){
	sample3 = sample2;
	}

	- buffer[sample * outputChannels + channel] = clamp<Size>(CubicInterpolate(input[sample0], input[sample1], input[sample2], input[sample3], inputSample - (int) inputSample));
	+ buffer[sample * outputChannels + channel] = clamp<SizeInput, SizeOutput>(CubicInterpolate(input[sample0], input[sample1], input[sample2], input[sample3], inputSample - (int) inputSample));

	// Global::debug(0) << "Input[" << sample << "] " << channel << ": " << input[sample1] << " Output: " << buffer[sample * 2 + channel] << std::endl;
	}
	}
	}

	int AudioConverter::convert(void * input, int length){
	/* no conversion needed */
	if (this->input == this->output){
	return length;
	}

	int total = convertedLength(length);
	/*
	if (total % byteSize(output) != 0){
	total -= total % byteSize(output);
	}
	*/

	/* cache the buffer for future use */
	if (total > bufferSize){
	delete[] buffer;
	bufferSize = total;
	buffer = new char[bufferSize];
	}

	double frequencyRatio = (double) output.frequency / (double) this->input.frequency;

	+ switch (this->input.bytes){
	+ case Signed16: {
	+ switch (this->output.bytes){
	+ case Signed16: doConvertRate<signed short, signed short>(
	+ (signed short*) input, length / sizeof(signed short) / this->input.channels, this->input.channels,
	+ (signed short*) buffer, total / sizeof(signed short) / output.channels, output.channels,
	+ frequencyRatio); break;
	+ case Unsigned8: doConvertRate<signed short, unsigned char>(
	+ (signed short*) input, length / sizeof(signed short) / this->input.channels, this->input.channels,
	+ (unsigned char*) buffer, total / sizeof(unsigned char) / output.channels, output.channels,
	+ frequencyRatio); break;
	+ case Unsigned16: doConvertRate<signed short, unsigned short>(
	+ (signed short*) input, length / sizeof(signed short) / this->input.channels, this->input.channels,
	+ (unsigned short*) buffer, total / sizeof(unsigned short) / output.channels, output.channels,
	+ frequencyRatio); break;
	+ case Float32: doConvertRate<signed short, float>(
	+ (signed short*) input, length / sizeof(signed short) / this->input.channels, this->input.channels,
	+ (float*) buffer, total / sizeof(float) / output.channels, output.channels,
	+ frequencyRatio); break;
	+
	+ }
	+ break;
	+ }
	+ case Float32: {
	+ switch (this->output.bytes){
	+ case Float32: doConvertRate<float, float>(
	+ (float*) input, length / sizeof(float) / this->input.channels, this->input.channels,
	+ (float*) buffer, total / sizeof(float) / output.channels, output.channels,
	+ frequencyRatio); break;
	+ default: break;
	+ }
	+ break;
	+ }
	+ default: break;
	+ }
	+
	+ /*
	if (this->input.bytes == output.bytes){
	switch (this->input.bytes){
	- case Signed16: doConvertRate<signed short>((signed short*) input, length / sizeof(signed short) / this->input.channels, this->input.channels,
	+ case Signed16: doConvertRate<signed short, signed short>((signed short*) input, length / sizeof(signed short) / this->input.channels, this->input.channels,
	(signed short*) buffer, total / sizeof(signed short) / output.channels, output.channels,
	frequencyRatio); break;
	- case Float32: doConvertRate<float>((float*) input, length / sizeof(float) / this->input.channels, this->input.channels,
	+ case Float32: doConvertRate<float, float>((float*) input, length / sizeof(float) / this->input.channels, this->input.channels,
	(float*) buffer, total / sizeof(float) / output.channels, output.channels, frequencyRatio); break;
	}
	}
	+ */

	memcpy(input, buffer, total);

	return total;
	}

	AudioConverter::~AudioConverter(){
	delete[] buffer;
	}

	bool AudioConverter::Format::operator==(const AudioConverter::Format & him) const {
	return this->bytes == him.bytes &&
	this->channels == him.channels &&
	this->frequency == him.frequency;
	}

	#endif

	}
	diff --git a/util/audio.h b/util/audio.h
	index 585adcb8..11694a8f 100644
	--- a/util/audio.h
	+++ b/util/audio.h
	@@ -1,63 +1,65 @@
	#ifndef _paintown_audio_h
	#define _paintown_audio_h

	#ifdef USE_SDL
	#include <SDL.h>
	#endif

	/* Deals with audio conversion between any source format and any destination format.
	* A format consists of
	* byte encoding (8/16/32 bit, signed/unsigned, floating point/integer)
	* number of channels (mono/stereo)
	* frequency (22050hz, 44100hz, arbitrary hz)
	*/

	namespace Util{

	/* endianness is always native */
	enum Encoding{
	+ Unsigned8,
	Signed16,
	+ Unsigned16,
	Float32
	};

	class AudioConverter{
	public:
	AudioConverter(Encoding inputEncoding, int inputChannels, int inputFrequency,
	Encoding outputEncoding, int outputChannels, int outputFrequency);

	/* given some input length, return how long the converted output will be */
	int convertedLength(int length);

	/* convert the audio, put the output in the same buffer passed in -- 'input'
	* and returns the number of converted samples.
	* 'length' is the number of input samples in bytes
	* 'input' should be large enough to hold convertedLength(length) samples
	*/
	int convert(void * input, int length);

	virtual ~AudioConverter();

	protected:
	#ifdef USE_SDL
	SDL_AudioCVT conversion;
	#endif
	struct Format{
	Encoding bytes;
	int channels;
	int frequency;

	bool operator==(const Format & him) const;
	};

	int byteSize(const Format & what);
	int encodingBytes(Encoding what);

	Format input, output;
	double sizeRatio;
	char * buffer;
	int bufferSize;
	};

	}

	#endif
	diff --git a/util/music-player.cpp b/util/music-player.cpp
	index ce2c6d5a..fc4ca076 100644
	--- a/util/music-player.cpp
	+++ b/util/music-player.cpp
	@@ -1,970 +1,971 @@
	#ifdef USE_ALLEGRO
	#include <allegro.h>
	#endif

	#include "music-player.h"
	#include "globals.h"
	#include "util/debug.h"
	#include <iostream>
	#include "configuration.h"
	#include "sound.h"
	#include "dumb/include/dumb.h"
	#include "gme/Music_Emu.h"
	#include "exceptions/exception.h"
	#include <sstream>
	#include <stdio.h>

	#ifdef USE_ALLEGRO5
	#include <allegro5/allegro_audio.h>
	#endif

	#ifdef USE_ALLEGRO
	#include "dumb/include/aldumb.h"

	#ifdef _WIN32
	/* what do we need winalleg for?
	* reason: ...
	*/
	#include <winalleg.h>
	#endif

	#endif

	#ifdef HAVE_MP3_MPG123
	#include <mpg123.h>
	#endif

	#ifdef HAVE_MP3_MAD
	#include <mad.h>
	#endif

	#ifdef USE_SDL
	#include "sdl/mixer/SDL_mixer.h"
	#endif

	using std::string;

	namespace Util{

	class MusicException: public Exception::Base {
	public:
	MusicException(const std::string & file, int line, const std::string & reason):
	Exception::Base(file, line),
	reason(reason){
	}

	MusicException(const MusicException & copy):
	Exception::Base(copy),
	reason(copy.reason){
	}

	virtual ~MusicException() throw(){
	}

	protected:
	virtual const std::string getReason() const {
	return reason;
	}

	virtual Exception::Base * copy() const {
	return new MusicException(*this);
	}

	std::string reason;
	};

	static double scaleVolume(double start){
	return start;
	}

	/* 1 for big endian (most significant byte)
	* 0 for little endian (least significant byte)
	*/
	/* FIXME: move this to global or something and find a better #ifdef */
	int bigEndian(){
	#if defined(PS3) \|\| defined(WII)
	return 1;
	#else
	return 0;
	#endif
	}

	#ifdef USE_ALLEGRO5
	const int DUMB_SAMPLES = 1024;
	MusicRenderer::MusicRenderer(){
	create(Sound::Info.frequency, 2);
	}

	MusicRenderer::MusicRenderer(int frequency, int channels){
	create(frequency, channels);
	}

	void MusicRenderer::create(int frequency, int channels){
	ALLEGRO_CHANNEL_CONF configuration = ALLEGRO_CHANNEL_CONF_2;
	switch (channels){
	case 1: configuration = ALLEGRO_CHANNEL_CONF_1; break;
	case 2: configuration = ALLEGRO_CHANNEL_CONF_2; break;
	case 3: configuration = ALLEGRO_CHANNEL_CONF_3; break;
	case 4: configuration = ALLEGRO_CHANNEL_CONF_4; break;
	case 5: configuration = ALLEGRO_CHANNEL_CONF_5_1; break;
	case 6: configuration = ALLEGRO_CHANNEL_CONF_6_1; break;
	case 7: configuration = ALLEGRO_CHANNEL_CONF_7_1; break;
	default: configuration = ALLEGRO_CHANNEL_CONF_2; break;
	}
	stream = al_create_audio_stream(4, DUMB_SAMPLES, frequency, ALLEGRO_AUDIO_DEPTH_INT16, configuration);
	if (!stream){
	throw MusicException(__FILE__, __LINE__, "Could not create allegro5 audio stream");
	}
	queue = al_create_event_queue();
	al_register_event_source(queue, al_get_audio_stream_event_source(stream));
	}

	void MusicRenderer::play(MusicPlayer & player){
	al_attach_audio_stream_to_mixer(stream, al_get_default_mixer());
	}

	void MusicRenderer::pause(){
	al_detach_audio_stream(stream);
	}

	MusicRenderer::~MusicRenderer(){
	al_destroy_audio_stream(stream);
	al_destroy_event_queue(queue);
	}

	void MusicRenderer::poll(MusicPlayer & player){
	ALLEGRO_EVENT event;
	while (al_get_next_event(queue, &event)){
	if (event.type == ALLEGRO_EVENT_AUDIO_STREAM_FRAGMENT) {
	ALLEGRO_AUDIO_STREAM * stream = (ALLEGRO_AUDIO_STREAM *) event.any.source;
	void * data = al_get_audio_stream_fragment(stream);
	if (data != NULL){
	player.render(data, al_get_audio_stream_length(stream));
	al_set_audio_stream_fragment(stream, data);
	}
	}
	}
	}
	#elif USE_SDL
	static const int BUFFER_SIZE = 1024 * 16;
	// static const int BUFFER_SIZE = 65536 * 2;
	Encoding formatType(int sdlFormat){
	switch (sdlFormat){
	+ case AUDIO_U8: return Unsigned8;
	case AUDIO_S16SYS: return Signed16;
	#if SDL_VERSION_ATLEAST(1, 3, 0)
	case AUDIO_F32MSB: return Float32;
	case AUDIO_F32LSB: return Float32;
	#endif
	}

	std::ostringstream out;
	out << "Don't know how to deal with SDL format " << sdlFormat << std::endl;
	throw MusicException(__FILE__, __LINE__, out.str());

	/*
	if (bigEndian()){
	switch (Sound::Info.format){
	case AUDIO_S16MSB: return Signed16;
	}
	return Signed16;
	} else {
	switch (Sound::Info.format){
	case AUDIO_S16LSB: return Signed16;
	}
	return Signed16;
	}
	*/
	}
	MusicRenderer::MusicRenderer():
	convert(formatType(AUDIO_S16SYS), Sound::Info.channels, Sound::Info.frequency,
	formatType(Sound::Info.format), Sound::Info.channels, Sound::Info.frequency){
	create(Sound::Info.frequency, Sound::Info.channels);
	}

	MusicRenderer::MusicRenderer(int frequency, int channels):
	convert(formatType(AUDIO_S16SYS), channels, frequency,
	formatType(Sound::Info.format), Sound::Info.channels, Sound::Info.frequency){
	create(frequency, channels);
	}

	void MusicRenderer::create(int frequency, int channels){

	// Global::debug(1) << "Convert between " << format << ", " << channels << ", " << frequency << " to " << Sound::Info.format << ", " << Sound::Info.channels << ", " << Sound::Info.frequency << std::endl;
	/*
	SDL_BuildAudioCVT(&convert, format, channels, frequency,
	Sound::Info.format, Sound::Info.channels,
	Sound::Info.frequency);
	*/
	int size = convert.convertedLength(BUFFER_SIZE);
	data = new Uint8[size < BUFFER_SIZE ? BUFFER_SIZE : size];
	position = 0;
	converted = 0;
	}

	static int sampleSize(){
	int size = 1;
	switch (Sound::Info.format){
	case AUDIO_U8:
	case AUDIO_S8: size = 1; break;
	case AUDIO_U16LSB:
	case AUDIO_S16LSB:
	case AUDIO_U16MSB:
	case AUDIO_S16MSB: size = 2; break;
	#if SDL_VERSION_ATLEAST(1, 3, 0)
	case AUDIO_S32LSB:
	case AUDIO_S32MSB:
	case AUDIO_F32LSB:
	case AUDIO_F32MSB: size = 2; break;
	#endif
	default: size = 2; break;
	}
	return size * Sound::Info.channels;
	}

	void MusicRenderer::fill(MusicPlayer * player){
	position = 0;
	/* read samples in dual-channel, 16-bit, signed form */
	player->render(data, BUFFER_SIZE / 4);
	converted = convert.convert(data, BUFFER_SIZE);
	/* sort of a hack, but we need exactly a multiple of 4 */
	/*
	int totalSample = sampleSize();
	if (converted % totalSample != 0){
	converted -= converted % totalSample;
	}
	*/
	// Global::debug(0) << "Filled " << converted << " bytes" << std::endl;
	#if 0
	if (convert.needed){
	convert.buf = data;
	convert.len = BUFFER_SIZE;
	/* then convert to whatever the real output wants */
	SDL_ConvertAudio(&convert);
	converted = convert.len_cvt;
	} else {
	converted = BUFFER_SIZE;
	}
	#endif
	}

	void MusicRenderer::read(MusicPlayer * player, Uint8 * stream, int bytes){
	// Global::debug(0) << "Need " << bytes << " bytes. Have " << (converted - position) << std::endl;
	while (bytes > 0){
	int length = bytes;
	if (length + position >= converted){
	length = converted - position;
	}
	/*
	if (length % 4 != 0){
	length -= length % 4;
	if (length == 0){
	fill(player);
	continue;
	}
	}
	*/

	// Global::debug(0) << "Copy " << length << " bytes" << std::endl;
	/* data contains samples in the same format as the output */
	memcpy(stream, data + position, length);
	stream += length;
	position += length;
	bytes -= length;
	if (position >= converted){
	fill(player);
	}
	}
	}

	void MusicRenderer::mixer(void * arg, Uint8 * stream, int bytes){
	MusicPlayer * player = (MusicPlayer*) arg;

	player->getRenderer()->read(player, stream, bytes);



	/*
	int size = (int)((float) bytes / player->getRenderer()->convert.len_ratio / (float) player->getRenderer()->convert.len_mult);
	Global::debug(2) << "Incoming " << bytes << " render " << size << std::endl;
	player->getRenderer()->convert.buf = player->getRenderer()->data;
	player->getRenderer()->convert.len = size;
	// player->render(stream, bytes / 4);
	player->render(player->getRenderer()->data, size / 4);

	SDL_ConvertAudio(&player->getRenderer()->convert);
	memcpy(stream, player->getRenderer()->data, bytes);
	*/
	}

	void MusicRenderer::play(MusicPlayer & player){
	Mix_HookMusic(mixer, &player);
	}

	void MusicRenderer::pause(){
	Mix_HookMusic(NULL, NULL);
	}

	void MusicRenderer::poll(MusicPlayer & player){
	}

	MusicRenderer::~MusicRenderer(){
	Mix_HookMusic(NULL, NULL);
	delete[] data;
	}
	#elif USE_ALLEGRO
	int BUFFER_SIZE = 1 << 11;
	static int ALLEGRO_MONO = 0;
	static int ALLEGRO_STEREO = 1;
	MusicRenderer::MusicRenderer(){
	create(Sound::Info.frequency, 2);
	}

	MusicRenderer::MusicRenderer(int frequency, int channels){
	create(frequency, channels);
	}

	void MusicRenderer::create(int frequency, int channels){
	int configuration = ALLEGRO_STEREO;
	if (channels == 1){
	configuration = ALLEGRO_MONO;
	}
	stream = play_audio_stream(BUFFER_SIZE, 16, configuration, frequency, 255, 128);
	if (!stream){
	throw MusicException(__FILE__, __LINE__, "Could not create Allegro stream");
	}
	if (stream->len != BUFFER_SIZE){
	throw MusicException(__FILE__, __LINE__, "Buffer size mismatch");
	}
	voice_set_priority(stream->voice, 255);
	}

	void MusicRenderer::play(MusicPlayer & player){
	voice_start(stream->voice);
	}

	void MusicRenderer::pause(){
	voice_stop(stream->voice);
	}

	void MusicRenderer::poll(MusicPlayer & player){
	short * buffer = (short*) get_audio_stream_buffer(stream);
	if (buffer){
	player.render(buffer, BUFFER_SIZE);

	/* allegro wants unsigned data but gme produces signed so to convert
	* signed samples to unsigned samples we have to raise each value
	* by half the maximum value of a short (0xffff+1)/2 = 0x8000
	*/
	for (int i = 0; i < BUFFER_SIZE * 2; i++){
	buffer[i] += 0x8000;
	}

	free_audio_stream_buffer(stream);
	}
	}

	MusicRenderer::~MusicRenderer(){
	stop_audio_stream(stream);
	}
	#endif

	MusicPlayer::MusicPlayer():
	volume(1.0),
	out(new MusicRenderer()){
	}

	MusicPlayer::~MusicPlayer(){
	}

	void MusicPlayer::setRenderer(const ReferenceCount<MusicRenderer> & what){
	this->out = what;
	}

	void MusicPlayer::play(){
	out->play(*this);
	}

	void MusicPlayer::pause(){
	out->pause();
	}

	void MusicPlayer::poll(){
	out->poll(*this);
	}

	static const char * typeToExtension( int i ){
	switch (i){
	case 0 : return ".xm";
	case 1 : return ".s3m";
	case 2 : return ".it";
	case 3 : return ".mod";
	default : return "";
	}
	}

	/* expects each sample to be 4 bytes, 2 bytes per sample * 2 channels */
	DumbPlayer::DumbPlayer(string path){
	music_file = loadDumbFile(path);
	if (music_file == NULL){
	std::ostringstream error;
	error << "Could not load DUMB file " << path;
	throw MusicException(__FILE__, __LINE__, error.str());
	}

	int n_channels = 2;
	int position = 0;
	renderer = duh_start_sigrenderer(music_file, 0, n_channels, position);
	if (!renderer){
	Global::debug(0) << "Could not create renderer" << std::endl;
	throw Exception::Base(__FILE__, __LINE__);
	}
	}

	void DumbPlayer::render(void * data, int samples){
	double delta = 65536.0 / Sound::Info.frequency;
	/* FIXME: use global music volume to scale the output here */
	int n = duh_render(renderer, 16, 0, volume, delta, samples, data);
	}

	void DumbPlayer::setVolume(double volume){
	this->volume = volume;
	}

	DumbPlayer::~DumbPlayer(){
	duh_end_sigrenderer(renderer);
	unload_duh(music_file);
	}

	DUH * DumbPlayer::loadDumbFile(string path){
	DUH * what;
	for (int i = 0; i < 4; i++){
	/* the order of trying xm/s3m/it/mod matters because mod could be
	* confused with one of the other formats, so load it last.
	*/
	switch (i){
	case 0 : {
	what = dumb_load_xm_quick(path.c_str());
	break;
	}
	case 1 : {
	what = dumb_load_s3m_quick(path.c_str());
	break;
	}
	case 2 : {
	what = dumb_load_it_quick(path.c_str());
	break;
	}
	case 3 : {
	what = dumb_load_mod_quick(path.c_str());
	break;
	}
	}

	if (what != NULL){
	Global::debug(0) << "Loaded " << path << " type " << typeToExtension(i) << "(" << i << ")" << std::endl;
	return what;
	}
	}
	return NULL;
	}

	GMEPlayer::GMEPlayer(string path):
	emulator(NULL){
	gme_err_t fail = gme_open_file(path.c_str(), &emulator, Sound::Info.frequency);
	if (fail != NULL){
	Global::debug(0) << "GME load error for " << path << ": " << fail << std::endl;
	throw MusicException(__FILE__, __LINE__, "Could not load GME file");
	}
	emulator->start_track(0);
	Global::debug(0) << "Loaded GME file " << path << std::endl;
	}

	void GMEPlayer::render(void * stream, int length){
	/* length/2 to convert bytes to short */
	emulator->play(length * 2, (short*) stream);

	if (emulator->track_ended()){
	gme_info_t * info;
	gme_track_info(emulator, &info, 0);
	int intro = info->intro_length;
	emulator->start_track(0);
	// Global::debug(0) << "Seeking " << intro << "ms. Track length " << info->length << "ms" << std::endl;
	/* skip past the intro if there is a loop */
	if (info->loop_length != 0){
	emulator->seek(intro);
	}
	}

	/* scale for volume */
	for (int i = 0; i < length * 2; i++){
	short & sample = ((short *) stream)[i];
	sample *= volume;
	}

	/*
	short large = 0;
	short small = 0;
	for (int i = 0; i < length / 2; i++){
	// ((short ) stream)[i] = 2;
	short z = ((short *) stream)[i];
	if (z < small){
	small = z;
	}
	if (z > large){
	large = z;
	}
	}
	Global::debug(0) << "Largest " << large << " Smallest " << small << std::endl;
	*/
	}

	void GMEPlayer::setVolume(double volume){
	this->volume = volume;
	}

	GMEPlayer::~GMEPlayer(){
	delete emulator;
	}

	#ifdef HAVE_MP3_MPG123
	/* initialize the mpg123 library and open up an mp3 file for reading */
	static void initializeMpg123(mpg123_handle ** mp3, string path){
	/* Initialize */
	if (mpg123_init() != MPG123_OK){
	throw MusicException(__FILE__, __LINE__, "Could not initialize mpg123");
	}
	try{
	*mp3 = mpg123_new(NULL, NULL);
	if (*mp3 == NULL){
	throw MusicException(__FILE__,__LINE__, "Could not allocate mpg handle");
	}
	mpg123_format_none(*mp3);

	/* allegro wants unsigned samples but mpg123 can't actually provide unsigned
	* samples even though it has an enum for it, MPG123_ENC_UNSIGNED_16. this
	* was rectified in 1.13.0 or something, but for now signed samples are ok.
	*/
	int error = mpg123_format(*mp3, Sound::Info.frequency, MPG123_STEREO, MPG123_ENC_SIGNED_16);
	if (error != MPG123_OK){
	Global::debug(0) << "Could not set format for mpg123 handle" << std::endl;
	}

	/* FIXME workaround for libmpg issues with "generic" decoder frequency not being set */
	error = mpg123_open(mp3, (char) path.c_str());
	if (error == -1){
	std::ostringstream error;
	error << "Could not open mpg123 file " << path << " error code " << error;
	throw MusicException(__FILE__,__LINE__, error.str());
	}

	/* reading a frame is the only surefire way to get mpg123 to set the
	* sampling_frequency which it needs to set the decoder a few lines below
	*/
	size_t dont_care;
	unsigned char tempBuffer[4096];
	error = mpg123_read(*mp3, tempBuffer, sizeof(tempBuffer), &dont_care);
	if (!(error == MPG123_OK \|\| error == MPG123_NEW_FORMAT)){
	std::ostringstream error;
	error << "Could not read mpg123 file " << path << " error code " << error;
	throw MusicException(__FILE__,__LINE__, error.str());
	}
	mpg123_close(*mp3);

	/* stream has progressed a little bit so reset it by opening it again */
	error = mpg123_open(mp3, (char) path.c_str());
	if (error == -1){
	std::ostringstream error;
	error << "Could not open mpg123 file " << path << " error code " << error;
	throw MusicException(__FILE__,__LINE__, error.str());
	}
	/* FIXME end */

	/* some of the native decoders aren't stable in older versions of mpg123
	* so just use generic for now. 1.13.1 should work better
	*/
	error = mpg123_decoder(*mp3, "generic");
	if (error != MPG123_OK){
	std::ostringstream error;
	error << "Could not use 'generic' mpg123 decoder for " << path << " error code " << error;
	throw MusicException(__FILE__,__LINE__, error.str());
	}
	// Global::debug(0) << "mpg support " << mpg123_format_support(mp3, Sound::FREQUENCY, MPG123_ENC_SIGNED_16) << std::endl;

	/*
	double base, really, rva;
	mpg123_getvolume(*mp3, &base, &really, &rva);
	// Global::debug(0) << "mpg volume base " << base << " really " << really << " rva " << rva << std::endl;
	base_volume = base;

	long rate;
	int channels, encoding;
	mpg123_getformat(*mp3, &rate, &channels, &encoding);
	// Global::debug(0) << path << " rate " << rate << " channels " << channels << " encoding " << encoding << std::endl;
	*/
	} catch (const MusicException & fail){
	if (*mp3 != NULL){
	mpg123_close(*mp3);
	mpg123_delete(*mp3);
	*mp3 = NULL;
	}
	mpg123_exit();
	throw;
	}
	}

	static const int MPG123_BUFFER_SIZE = 1 << 11;
	Mp3Player::Mp3Player(string path):
	mp3(NULL){
	initializeMpg123(&mp3, path);
	long rate = 0;
	int channels = 0, encoding = 0;
	mpg123_getformat(mp3, &rate, &channels, &encoding);
	}

	void Mp3Player::render(void * data, int samples){
	/* buffer * 4 for 16 bits per sample * 2 samples for stereo */
	size_t out = 0;
	mpg123_read(mp3, (unsigned char ) data, samples 4, &out);

	/*
	long rate;
	int channels, encoding;
	mpg123_getformat(mp3, &rate, &channels, &encoding);
	Global::debug(0) << "rate " << rate << " channels " << channels << " encoding " << encoding << std::endl;
	*/
	}

	void Mp3Player::setVolume(double volume){
	mpg123_volume(mp3, volume);
	/*
	this->volume = volume;
	// mpg123_volume(mp3, volume * base_volume / 5000);
	mpg123_volume(mp3, 0.0001);
	*/
	// mpg123_volume(mp3, volume);
	}

	Mp3Player::~Mp3Player(){
	mpg123_close(mp3);
	mpg123_exit();
	}

	#endif /* MP3_MPG123 */

	#ifdef HAVE_OGG
	int OGG_BUFFER_SIZE = 1024 * 32;
	OggPlayer::OggPlayer(string path):
	path(path){
	file = fopen(path.c_str(), "rb");
	if (!file) {
	throw MusicException(__FILE__, __LINE__, "Could not open file");
	}

	if (ov_open_callbacks(file, &ogg, 0, 0, OV_CALLBACKS_DEFAULT) != 0) {
	fclose(file);
	throw MusicException(__FILE__, __LINE__, "Could not open ogg");
	}

	vorbis_info * info = ov_info(&ogg, -1);

	frequency = info->rate;
	channels = info->channels;
	bits = 16;
	length = ov_pcm_total(&ogg, -1);

	setRenderer(new MusicRenderer(info->rate, info->channels));

	buffer = new OggPage();
	buffer->buffer1.buffer = new char[OGG_BUFFER_SIZE];
	// buffer->buffer2.buffer = new char[OGG_BUFFER_SIZE];

	fillPage(&buffer->buffer1);
	// fillPage(&buffer->buffer2);
	// buffer->use = 0;
	}

	void OggPlayer::fillPage(OggPage::Page * page){
	int dont_care;
	page->position = 0;
	page->max = 0;
	while (page->max < OGG_BUFFER_SIZE){
	/* ov_read might not read all available samples, I guess it stops
	* reading on a page boundary. We just plow on through.
	*/
	int read = ov_read(&ogg, (char*) page->buffer + page->max, OGG_BUFFER_SIZE - page->max,
	bigEndian(), 2, 1, &dont_care);
	/* if we hit the end of the file then re-open it and keep reading */
	if (read == 0){
	ov_clear(&ogg);
	file = fopen(path.c_str(), "rb");
	if (!file){
	throw MusicException(__FILE__, __LINE__, "Could not open file");
	}
	int ok = ov_open_callbacks(file, &ogg, 0, 0, OV_CALLBACKS_DEFAULT);
	if (ok != 0){
	fclose(file);
	throw MusicException(__FILE__, __LINE__, "Could not open ogg");
	}
	} else if (read == OV_HOLE){
	throw MusicException(__FILE__, __LINE__, "Garbage in ogg file");
	} else if (read == OV_EBADLINK){
	throw MusicException(__FILE__, __LINE__, "Invalid stream section in ogg");
	} else if (read == OV_EINVAL){
	throw MusicException(__FILE__, __LINE__, "File headers are corrupt in ogg");
	} else {
	page->max += read;
	}
	}
	}

	void OggPlayer::doRender(char * data, int bytes){
	OggPage::Page & page = buffer->buffer1;
	if (page.max - page.position >= bytes){
	memcpy(data, page.buffer + page.position, bytes);
	page.position += bytes;
	} else {
	/* copy the rest, fill the page, switch to the other buffer */
	memcpy(data, page.buffer + page.position, page.max - page.position);
	int at = page.max - page.position;
	int rest = bytes - (page.max - page.position);
	fillPage(&page);
	doRender(data + at, rest);
	}
	}

	void OggPlayer::render(void * data, int length){
	doRender((char) data, length 4);
	}

	void OggPlayer::setVolume(double volume){
	this->volume = volume;
	// Mix_VolumeMusic(volume * MIX_MAX_VOLUME);
	}

	OggPlayer::~OggPlayer(){
	/* ov_clear will close the file */
	ov_clear(&ogg);
	}
	#endif /* OGG */

	#ifdef HAVE_MP3_MAD
	Mp3Player::Mp3Player(string path):
	available(NULL),
	bytesLeft(0),
	position(0),
	raw(NULL){
	FILE * handle = fopen(path.c_str(), "rb");
	if (!handle){
	std::ostringstream out;
	out << "Could not open mp3 file " << path;
	throw MusicException(__FILE__, __LINE__, out.str());
	}

	fseek(handle, 0, SEEK_END);
	rawLength = ftell(handle);
	fseek(handle, 0, SEEK_SET);
	raw = new unsigned char[rawLength];

	int toRead = rawLength;
	int where =0;
	while (toRead > 0){
	int got = fread(raw + where, 1, toRead, handle);
	toRead -= got;
	}
	fclose(handle);

	int rate = 44100, channels = 2;
	discoverInfo(raw, rawLength, &rate, &channels);
	setRenderer(new MusicRenderer(rate, channels));

	Global::debug(0) << "Opened mp3 file " << path << " rate " << rate << " channels " << channels << std::endl;

	mad_stream_init(&stream);
	mad_frame_init(&frame);
	mad_synth_init(&synth);
	mad_stream_buffer(&stream, raw, rawLength);

	fill(4);
	}

	/* read the first frame and get the rate and channels from the header.
	* assume all other frames use the same rate and channels
	*/
	void Mp3Player::discoverInfo(unsigned char * raw, int length, int * rate, int * channels){
	mad_frame frame;
	mad_stream stream;
	mad_frame_init(&frame);
	mad_stream_init(&stream);
	mad_stream_buffer(&stream, raw, length);
	int ok = mad_header_decode(&frame.header, &stream);
	while (ok == -1){
	if (MAD_RECOVERABLE(stream.error)){
	ok = mad_header_decode(&frame.header, &stream);
	} else {
	throw MusicException(__FILE__, __LINE__, "Could not decode mp3 frame");
	}
	}
	*rate = frame.header.samplerate;
	switch (frame.header.mode){
	case MAD_MODE_SINGLE_CHANNEL: *channels = 1; break;
	case MAD_MODE_DUAL_CHANNEL: *channels = 2; break;
	case MAD_MODE_JOINT_STEREO: *channels = 2; break;
	case MAD_MODE_STEREO: *channels = 2; break;
	}

	mad_frame_finish(&frame);
	mad_stream_finish(&stream);
	}

	mad_flow Mp3Player::error(void * data, mad_stream * stream, mad_frame * frame){
	if (MAD_RECOVERABLE(stream->error)){
	return MAD_FLOW_CONTINUE;
	}
	throw MusicException(__FILE__, __LINE__, "Error decoding mp3 stream");
	}

	static inline signed int mad_scale(mad_fixed_t sample){
	/* round */
	sample += (1L << (MAD_F_FRACBITS - 16));

	/* clip */
	if (sample >= MAD_F_ONE)
	sample = MAD_F_ONE - 1;
	else if (sample < -MAD_F_ONE)
	sample = -MAD_F_ONE;

	/* quantize */
	return sample >> (MAD_F_FRACBITS + 1 - 16);
	}

	void Mp3Player::output(mad_header const * header, mad_pcm * pcm){
	unsigned int channels = pcm->channels;
	unsigned int samples = pcm->length;
	/*
	mad_fixed_t const * left = pcm->samples[0];
	mad_fixed_t const * right = pcm->samples[1];
	*/

	unsigned short * out = new unsigned short[samples * channels];
	for (unsigned int index = 0; index < samples; index++){
	for (int channel = 0; channel < channels; channel++){
	mad_fixed_t const * left = pcm->samples[channel] + index;
	out[index * channels + channel] = mad_scale(*left) & 0xffff;
	// out[index * 2 + 1] = mad_scale(*right) & 0xffff;
	}
	// left += 1;
	// right += 1;
	}

	/* N channels * 2 bytes per sample */
	pages.push_back(Data((char) out, samples channels * 2));
	}

	mad_flow Mp3Player::input(void * data, mad_stream * stream){
	/*
	Mp3Player * player = (Mp3Player*) data;
	if (!player->readMore){
	return MAD_FLOW_STOP;
	} else {
	player->readMore = false;
	}
	int read = fread(player->raw, 1, RAW_SIZE, player->handle);
	if (feof(player->handle)){
	/ * start over * /
	fseek(player->handle, 0, SEEK_SET);
	}
	mad_stream_buffer(stream, player->raw, read);
	return MAD_FLOW_CONTINUE;
	*/
	return MAD_FLOW_CONTINUE;
	}

	void Mp3Player::fill(int frames){
	for (int i = 0; i < frames; i++){
	int headerError = mad_header_decode(&frame.header, &stream);
	while (headerError == -1){
	if (MAD_RECOVERABLE(stream.error)){
	} else {
	if (stream.error == MAD_ERROR_BUFLEN){
	mad_stream_finish(&stream);
	mad_frame_finish(&frame);
	mad_synth_finish(&synth);

	mad_stream_init(&stream);
	mad_frame_init(&frame);
	mad_synth_init(&synth);
	mad_stream_buffer(&stream, raw, rawLength);
	}
	}
	headerError = mad_header_decode(&frame.header, &stream);
	}

	mad_frame_decode(&frame, &stream);
	mad_synth_frame(&synth, &frame);
	output(&frame.header, &synth.pcm);
	}

	/*
	readMore = true;
	int result = mad_decoder_run(&decoder, MAD_DECODER_MODE_SYNC);
	*/

	bytesLeft = 0;
	for (std::vector<Data>::iterator it = pages.begin(); it != pages.end(); it++){
	bytesLeft += it->length;
	}

	// Global::debug(0) << "Read " << bytesLeft << std::endl;

	delete[] available;
	available = new char[bytesLeft];
	position = 0;
	int here = 0;
	for (std::vector<Data>::iterator it = pages.begin(); it != pages.end(); it++){
	memcpy(available + here, it->data, it->length);
	here += it->length;

	delete[] it->data;
	}

	// Global::debug(0) << "Filled mp3 with " << bytesLeft << std::endl;

	pages.clear();
	}

	void Mp3Player::render(void * data, int length){
	length *= 4;
	// Global::debug(0) << "Mp3 render " << length << " have " << bytesLeft << std::endl;
	while (length > 0){
	int left = length;
	if (left > bytesLeft){
	left = bytesLeft;
	}
	memcpy(data, available + position, left);
	length -= left;
	bytesLeft -= left;
	position += left;
	data = ((char*) data) + left;

	if (bytesLeft == 0){
	fill(4);
	}
	}
	}

	void Mp3Player::setVolume(double volume){
	/* TODO */
	}

	Mp3Player::~Mp3Player(){
	delete[] raw;
	delete[] available;
	mad_stream_finish(&stream);
	mad_frame_finish(&frame);
	mad_synth_finish(&synth);
	// mad_decoder_finish(&decoder);
	}
	#endif /* MP3_MAD */


	}
	diff --git a/util/sdl/sound.cpp b/util/sdl/sound.cpp
	index c467704d..e090f75f 100644
	--- a/util/sdl/sound.cpp
	+++ b/util/sdl/sound.cpp
	@@ -1,95 +1,98 @@
	#include "../sound.h"
	#include <SDL.h>
	#include "mixer/SDL_mixer.h"
	+#include "util/debug.h"

	Sound::Sound():
	own(NULL){
	}

	/* create from wav file (riff header + pcm) */
	Sound::Sound(const char * data, int length):
	own(NULL){
	SDL_RWops * ops = SDL_RWFromConstMem(data, length);
	this->data.chunk = Mix_LoadWAV_RW(ops, 1);
	own = new int;
	*own = 1;
	}

	/* load from path */
	Sound::Sound(const std::string & path) throw (LoadException):
	own(NULL){
	data.chunk = Mix_LoadWAV(path.c_str());
	if (!data.chunk){
	printf("Can't load sound %s\n", path.c_str());
	// throw LoadException("Could not load sound " + path);
	} else {
	own = new int;
	*own = 1;
	}
	}

	void Sound::initialize(){
	// int audio_rate = 22050;
	/* allegro uses 44100 by default with alsa9 */
	int audio_rate = Info.frequency;
	// int audio_rate = 22050;
	Uint16 audio_format = AUDIO_S16;
	// Uint16 audio_format = MIX_DEFAULT_FORMAT;
	int audio_channels = 2;
	int audio_buffers = 4096;
	// int audio_buffers = 44100;
	if (Mix_OpenAudio(audio_rate, audio_format, audio_channels, audio_buffers)) {
	printf("Unable to open audio: %s!\n", Mix_GetError());
	// exit(1);
	}

	/* use the frequency enforced by the audio system */
	Mix_QuerySpec(&audio_rate, &audio_format, &audio_channels);
	Info.frequency = audio_rate;
	Info.channels = audio_channels;
	Info.format = audio_format;
	+
	+ Global::debug(0) << "Opened audio at rate " << audio_rate << " channels " << audio_channels << " format " << audio_format << std::endl;
	}

	void Sound::uninitialize(){
	Mix_CloseAudio();
	}

	void Sound::play(){
	if (data.chunk != NULL){
	Mix_VolumeChunk(data.chunk, (int) scale(MIX_MAX_VOLUME));
	Mix_PlayChannel(-1, data.chunk, 0);
	}
	}

	void Sound::play(double volume, int pan){
	if (data.chunk != NULL){
	Mix_VolumeChunk(data.chunk, (int) scale(volume * MIX_MAX_VOLUME));
	Mix_PlayChannel(-1, data.chunk, 0);
	}
	}

	void Sound::playLoop(){
	if (data.chunk != NULL){
	Mix_VolumeChunk(data.chunk, (int) scale(MIX_MAX_VOLUME));
	Mix_PlayChannel(-1, data.chunk, -1);
	}
	}

	void Sound::destroy(){
	if (own){
	*own -= 1;
	if ( *own == 0 ){
	delete own;
	if (data.chunk != NULL){
	Mix_FreeChunk(data.chunk);
	}
	own = NULL;
	}
	}
	}

	void Sound::stop(){
	if (data.channel != -1){
	Mix_HaltChannel(data.channel);
	}
	}

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