AUDIO: Rename sound/ dir to audio/

svn-id: r55850

16 files changed, 5607 insertions, 0 deletions

diff --git a/audio/mods/infogrames.cpp b/audio/mods/infogrames.cpp
new file mode 100644
index 0000000000..27e42c637b
--- /dev/null
+++ b/audio/mods/infogrames.cpp
@@ -0,0 +1,470 @@
+/* ScummVM - Graphic Adventure Engine
+ *
+ * ScummVM is the legal property of its developers, whose names
+ * are too numerous to list here. Please refer to the COPYRIGHT
+ * file distributed with this source distribution.
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
+ *
+ * $URL$
+ * $Id$
+ *
+ */
+
+#include "audio/mods/infogrames.h"
+#include "common/endian.h"
+#include "common/file.h"
+#include "common/memstream.h"
+
+namespace Audio {
+
+Infogrames::Instruments::Instruments() {
+	init();
+}
+
+Infogrames::Instruments::~Instruments() {
+	delete[] _sampleData;
+}
+
+void Infogrames::Instruments::init() {
+	int i;
+
+	for (i = 0; i < 32; i++) {
+		_samples[i].data = 0;
+		_samples[i].dataRepeat = 0;
+		_samples[i].length = 0;
+		_samples[i].lengthRepeat = 0;
+	}
+	_count = 0;
+	_sampleData = 0;
+}
+
+bool Infogrames::Instruments::load(const char *ins) {
+	Common::File f;
+
+	if (f.open(ins))
+		return load(f);
+	return false;
+}
+
+bool Infogrames::Instruments::load(Common::SeekableReadStream &ins) {
+	int i;
+	int32 fsize;
+	int32 offset[32];
+	int32 offsetRepeat[32];
+	int32 dataOffset;
+
+	unload();
+
+	fsize = ins.readUint32BE();
+	dataOffset = fsize;
+	for (i = 0; (i < 32) && !ins.eos(); i++) {
+		offset[i] = ins.readUint32BE();
+		offsetRepeat[i] = ins.readUint32BE();
+		if ((offset[i] > fsize) || (offsetRepeat[i] > fsize) ||
+				(offset[i] < (ins.pos() + 4)) ||
+				(offsetRepeat[i] < (ins.pos() + 4))) {
+			// Definitely no real entry anymore
+			ins.seek(-8, SEEK_CUR);
+			break;
+		}
+
+		dataOffset = MIN(dataOffset, MIN(offset[i], offsetRepeat[i]));
+		ins.skip(4); // Unknown
+		_samples[i].length = ins.readUint16BE() * 2;
+		_samples[i].lengthRepeat = ins.readUint16BE() * 2;
+	}
+
+	if (dataOffset >= fsize)
+		return false;
+
+	_count = i;
+	_sampleData = new int8[fsize - dataOffset];
+	ins.seek(dataOffset + 4);
+	ins.read(_sampleData, fsize - dataOffset);
+
+	for (i--; i >= 0; i--) {
+		_samples[i].data = _sampleData + (offset[i] - dataOffset);
+		_samples[i].dataRepeat = _sampleData + (offsetRepeat[i] - dataOffset);
+	}
+
+	return true;
+}
+
+void Infogrames::Instruments::unload() {
+	delete[] _sampleData;
+	init();
+}
+
+const uint8 Infogrames::tickCount[] =
+	{2, 3, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96};
+const uint16 Infogrames::periods[] =
+	{0x6ACC, 0x64CC, 0x5F25, 0x59CE, 0x54C3, 0x5003, 0x4B86, 0x4747, 0x4346,
+	 0x3F8B, 0x3BF3, 0x3892, 0x3568, 0x3269, 0x2F93, 0x2CEA, 0x2A66, 0x2801,
+	 0x2566, 0x23A5, 0x21AF, 0x1FC4, 0x1DFE, 0x1C4E, 0x1ABC, 0x1936, 0x17CC,
+	 0x1676, 0x1533, 0x1401, 0x12E4, 0x11D5, 0x10D4, 0x0FE3, 0x0EFE, 0x0E26,
+	 0x0D5B, 0x0C9B, 0x0BE5, 0x0B3B, 0x0A9B, 0x0A02, 0x0972, 0x08E9, 0x0869,
+	 0x07F1, 0x077F, 0x0713, 0x06AD, 0x064D, 0x05F2, 0x059D, 0x054D, 0x0500,
+	 0x04B8, 0x0475, 0x0435, 0x03F8, 0x03BF, 0x038A, 0x0356, 0x0326, 0x02F9,
+	 0x02CF, 0x02A6, 0x0280, 0x025C, 0x023A, 0x021A, 0x01FC, 0x01E0, 0x01C5,
+	 0x01AB, 0x0193, 0x017D, 0x0167, 0x0153, 0x0140, 0x012E, 0x011D, 0x010D,
+	 0x00FE, 0x00F0, 0x00E2, 0x00D6, 0x00CA, 0x00BE, 0x00B4, 0x00AA, 0x00A0,
+	 0x0097, 0x008F, 0x0087, 0x007F, 0x0078, 0x0070, 0x0060, 0x0050, 0x0040,
+	 0x0030, 0x0020, 0x0010, 0x0000, 0x0000, 0x0020, 0x2020, 0x2020, 0x2020,
+	 0x2020, 0x3030, 0x3030, 0x3020, 0x2020, 0x2020, 0x2020, 0x2020, 0x2020,
+	 0x2020, 0x2020, 0x2020, 0x2090, 0x4040, 0x4040, 0x4040, 0x4040, 0x4040,
+	 0x4040, 0x4040, 0x400C, 0x0C0C, 0x0C0C, 0x0C0C, 0x0C0C, 0x0C40, 0x4040,
+	 0x4040, 0x4040, 0x0909, 0x0909, 0x0909, 0x0101, 0x0101, 0x0101, 0x0101,
+	 0x0101, 0x0101, 0x0101, 0x0101, 0x0101, 0x0101, 0x4040, 0x4040, 0x4040,
+	 0x0A0A, 0x0A0A, 0x0A0A, 0x0202, 0x0202, 0x0202, 0x0202, 0x0202, 0x0202,
+	 0x0202, 0x0202, 0x0202, 0x0202, 0x4040, 0x4040, 0x2000};
+
+Infogrames::Infogrames(Instruments &ins, bool stereo, int rate,
+		int interruptFreq) : Paula(stereo, rate, interruptFreq) {
+	_instruments = &ins;
+	_data = 0;
+	_repCount = -1;
+
+	reset();
+}
+
+Infogrames::~Infogrames() {
+	delete[] _data;
+}
+
+void Infogrames::init() {
+	int i;
+
+	_volume = 0;
+	_period = 0;
+	_sample = 0;
+	_speedCounter = _speed;
+
+	for (i = 0; i < 4; i++) {
+		_chn[i].cmds = 0;
+		_chn[i].cmdBlocks = 0;
+		_chn[i].volSlide.finetuneNeg = 0;
+		_chn[i].volSlide.finetunePos = 0;
+		_chn[i].volSlide.data = 0;
+		_chn[i].volSlide.amount = 0;
+		_chn[i].volSlide.dataOffset = 0;
+		_chn[i].volSlide.flags = 0;
+		_chn[i].volSlide.curDelay1 = 0;
+		_chn[i].volSlide.curDelay2 = 0;
+		_chn[i].periodSlide.finetuneNeg = 0;
+		_chn[i].periodSlide.finetunePos = 0;
+		_chn[i].periodSlide.data = 0;
+		_chn[i].periodSlide.amount = 0;
+		_chn[i].periodSlide.dataOffset = 0;
+		_chn[i].periodSlide.flags = 0;
+		_chn[i].periodSlide.curDelay1 = 0;
+		_chn[i].periodSlide.curDelay2 = 0;
+		_chn[i].period = 0;
+		_chn[i].flags = 0x81;
+		_chn[i].ticks = 0;
+		_chn[i].tickCount = 0;
+		_chn[i].periodMod = 0;
+	}
+
+	_end = (_data == 0);
+}
+
+void Infogrames::reset() {
+	int i;
+
+	stopPlay();
+	init();
+
+	_volSlideBlocks = 0;
+	_periodSlideBlocks = 0;
+	_subSong = 0;
+	_cmdBlocks = 0;
+	_speedCounter = 0;
+	_speed = 0;
+
+	for (i = 0; i < 4; i++)
+		_chn[i].cmdBlockIndices = 0;
+}
+
+bool Infogrames::load(const char *dum) {
+	Common::File f;
+
+	if (f.open(dum))
+		return load(f);
+	return false;
+}
+
+bool Infogrames::load(Common::SeekableReadStream &dum) {
+	int subSong = 0;
+	int i;
+	uint32 size;
+
+	size = dum.size();
+	if (size < 20)
+		return false;
+
+	_data = new uint8[size];
+	dum.seek(0);
+	dum.read(_data, size);
+
+	Common::MemoryReadStream dataStr(_data, size);
+
+	dataStr.seek(subSong * 2);
+	dataStr.seek(dataStr.readUint16BE());
+	_subSong = _data + dataStr.pos();
+	if (_subSong > (_data + size))
+		return false;
+
+	_speedCounter = dataStr.readUint16BE();
+	_speed = _speedCounter;
+	_volSlideBlocks = _subSong + dataStr.readUint16BE();
+	_periodSlideBlocks = _subSong + dataStr.readUint16BE();
+	for (i = 0; i < 4; i++) {
+		_chn[i].cmdBlockIndices = _subSong + dataStr.readUint16BE();
+		_chn[i].flags = 0x81;
+	}
+	_cmdBlocks = _data + dataStr.pos() + 2;
+
+	if ((_volSlideBlocks > (_data + size)) ||
+			(_periodSlideBlocks > (_data + size)) ||
+			(_chn[0].cmdBlockIndices > (_data + size)) ||
+			(_chn[1].cmdBlockIndices > (_data + size)) ||
+			(_chn[2].cmdBlockIndices > (_data + size)) ||
+			(_chn[3].cmdBlockIndices > (_data + size)) ||
+			(_cmdBlocks > (_data + size)))
+		return false;
+
+	startPaula();
+	return true;
+}
+
+void Infogrames::unload() {
+	stopPlay();
+
+	delete[] _data;
+	_data = 0;
+
+	clearVoices();
+	reset();
+}
+
+void Infogrames::getNextSample(Channel &chn) {
+	byte *data;
+	byte cmdBlock = 0;
+	uint16 cmd;
+	bool cont = false;
+
+	if (chn.flags & 64)
+		return;
+
+	if (chn.flags & 1) {
+		chn.flags &= ~1;
+		chn.cmdBlocks = chn.cmdBlockIndices;
+	} else {
+		chn.flags &= ~1;
+		if (_speedCounter == 0)
+			chn.ticks--;
+		if (chn.ticks != 0) {
+			_volume = MAX((int16) 0, tune(chn.volSlide, 0));
+			_period = tune(chn.periodSlide, chn.period);
+			return;
+		} else {
+			chn.ticks = chn.tickCount;
+			cont = true;
+		}
+	}
+
+	while (1) {
+		while (cont || ((cmdBlock = *chn.cmdBlocks) != 0xFF)) {
+			if (!cont) {
+				chn.cmdBlocks++;
+				chn.cmds = _subSong +
+					READ_BE_UINT16(_cmdBlocks + (cmdBlock * 2));
+			} else
+				cont = false;
+			while ((cmd = *chn.cmds) != 0xFF) {
+				chn.cmds++;
+				if (cmd & 128)
+				{
+					switch (cmd & 0xE0) {
+					case 0x80: // 100xxxxx - Set ticks
+						chn.ticks = tickCount[cmd & 0xF];
+						chn.tickCount = tickCount[cmd & 0xF];
+						break;
+					case 0xA0: // 101xxxxx - Set sample
+						_sample = cmd & 0x1F;
+						break;
+					case 0xC0: // 110xxxxx - Set volume slide/finetune
+						data = _volSlideBlocks + (cmd & 0x1F) * 13;
+						chn.volSlide.flags = (*data & 0x80) | 1;
+						chn.volSlide.amount = *data++ & 0x7F;
+						chn.volSlide.data = data;
+						chn.volSlide.dataOffset = 0;
+						chn.volSlide.finetunePos = 0;
+						chn.volSlide.finetuneNeg = 0;
+						chn.volSlide.curDelay1 = 0;
+						chn.volSlide.curDelay2 = 0;
+						break;
+					case 0xE0: // 111xxxxx - Extended
+						switch (cmd & 0x1F) {
+						case 0: // Set period modifier
+							chn.periodMod = (int8) *chn.cmds++;
+							break;
+						case 1: // Set continuous period slide
+							chn.periodSlide.data =
+								_periodSlideBlocks + *chn.cmds++ * 13 + 1;
+							chn.periodSlide.amount = 0;
+							chn.periodSlide.dataOffset = 0;
+							chn.periodSlide.finetunePos = 0;
+							chn.periodSlide.finetuneNeg = 0;
+							chn.periodSlide.curDelay1 = 0;
+							chn.periodSlide.curDelay2 = 0;
+							chn.periodSlide.flags = 0x81;
+							break;
+						case 2: // Set non-continuous period slide
+							chn.periodSlide.data =
+								_periodSlideBlocks + *chn.cmds++ * 13 + 1;
+							chn.periodSlide.amount = 0;
+							chn.periodSlide.dataOffset = 0;
+							chn.periodSlide.finetunePos = 0;
+							chn.periodSlide.finetuneNeg = 0;
+							chn.periodSlide.curDelay1 = 0;
+							chn.periodSlide.curDelay2 = 0;
+							chn.periodSlide.flags = 1;
+							break;
+						case 3: // NOP
+							break;
+						default:
+							warning("Unknown Infogrames command: %X", cmd);
+						}
+						break;
+					}
+				} else { // 0xxxxxxx - Set period
+					if (cmd != 0)
+						cmd += chn.periodMod;
+					chn.period = periods[cmd];
+					chn.volSlide.dataOffset = 0;
+					chn.volSlide.finetunePos = 0;
+					chn.volSlide.finetuneNeg = 0;
+					chn.volSlide.curDelay1 = 0;
+					chn.volSlide.curDelay2 = 0;
+					chn.volSlide.flags |= 1;
+					chn.volSlide.flags &= ~4;
+					chn.periodSlide.dataOffset = 0;
+					chn.periodSlide.finetunePos = 0;
+					chn.periodSlide.finetuneNeg = 0;
+					chn.periodSlide.curDelay1 = 0;
+					chn.periodSlide.curDelay2 = 0;
+					chn.periodSlide.flags |= 1;
+					chn.periodSlide.flags &= ~4;
+					_volume = MAX((int16) 0, tune(chn.volSlide, 0));
+					_period = tune(chn.periodSlide, chn.period);
+					return;
+				}
+			}
+		}
+		if (!(chn.flags & 32)) {
+			chn.flags |= 0x40;
+			_volume = 0;
+			return;
+		} else
+			chn.cmdBlocks = chn.cmdBlockIndices;
+	}
+}
+
+int16 Infogrames::tune(Slide &slide, int16 start) const {
+	byte *data;
+	uint8 off;
+
+	data = slide.data + slide.dataOffset;
+
+	if (slide.flags & 1)
+		slide.finetunePos += (int8) data[1];
+	slide.flags &= ~1;
+
+	start += slide.finetunePos - slide.finetuneNeg;
+	if (start < 0)
+		start = 0;
+
+	if (slide.flags & 4)
+		return start;
+
+	slide.curDelay1++;
+	if (slide.curDelay1 != data[2])
+		return start;
+	slide.curDelay2++;
+	slide.curDelay1 = 0;
+	if (slide.curDelay2 == data[0]) {
+		slide.curDelay2 = 0;
+		off = slide.dataOffset + 3;
+		if (off == 12) {
+			if (slide.flags == 0) {
+				slide.flags |= 4;
+				return start;
+			} else {
+				slide.curDelay2 = 0;
+				slide.finetuneNeg += slide.amount;
+				off = 3;
+			}
+		}
+		slide.dataOffset = off;
+	}
+	slide.flags |= 1;
+	return start;
+}
+
+void Infogrames::interrupt() {
+	int chn;
+
+	if (!_data) {
+		clearVoices();
+		return;
+	}
+
+	_speedCounter--;
+	_sample = 0xFF;
+	for (chn = 0; chn < 4; chn++) {
+		_volume = 0;
+		_period = 0;
+		getNextSample(_chn[chn]);
+		setChannelVolume(chn, _volume);
+		setChannelPeriod(chn, _period);
+		if ((_sample != 0xFF) && (_sample < _instruments->_count)) {
+			setChannelData(chn,
+			               _instruments->_samples[_sample].data,
+			               _instruments->_samples[_sample].dataRepeat,
+			               _instruments->_samples[_sample].length,
+			               _instruments->_samples[_sample].lengthRepeat);
+			_sample = 0xFF;
+		}
+	}
+	if (_speedCounter == 0)
+		_speedCounter = _speed;
+
+	// End reached?
+	if ((_chn[0].flags & 64) && (_chn[1].flags & 64) &&
+			(_chn[2].flags & 64) && (_chn[3].flags & 64)) {
+		if (_repCount > 0) {
+			_repCount--;
+			init();
+		} else if (_repCount != -1) {
+			stopPaula();
+		} else {
+			init();
+		}
+	}
+}
+
+} // End of namespace Audio
diff --git a/audio/mods/infogrames.h b/audio/mods/infogrames.h
new file mode 100644
index 0000000000..c7abebf24e
--- /dev/null
+++ b/audio/mods/infogrames.h
@@ -0,0 +1,148 @@
+/* ScummVM - Graphic Adventure Engine
+ *
+ * ScummVM is the legal property of its developers, whose names
+ * are too numerous to list here. Please refer to the COPYRIGHT
+ * file distributed with this source distribution.
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
+ *
+ * $URL$
+ * $Id$
+ *
+ */
+
+/**
+ * @file
+ * Sound decoder used in engines:
+ *  - gob
+ */
+
+#ifndef SOUND_MODS_INFOGRAMES_H
+#define SOUND_MODS_INFOGRAMES_H
+
+#include "audio/mods/paula.h"
+#include "common/stream.h"
+
+namespace Audio {
+
+/** A player for the Infogrames/RobHubbard2 format */
+class Infogrames : public Paula {
+public:
+	class Instruments {
+	public:
+		Instruments();
+		template<typename T> Instruments(T ins) {
+			init();
+			bool result = load(ins);
+			assert(result);
+		}
+		~Instruments();
+
+		bool load(Common::SeekableReadStream &ins);
+		bool load(const char *ins);
+		void unload();
+
+		uint8 getCount() const { return _count; }
+
+	protected:
+		struct Sample {
+			int8 *data;
+			int8 *dataRepeat;
+			uint32 length;
+			uint32 lengthRepeat;
+		} _samples[32];
+
+		uint8 _count;
+		int8 *_sampleData;
+
+		void init();
+
+		friend class Infogrames;
+	};
+
+	Infogrames(Instruments &ins, bool stereo = false, int rate = 44100,
+			int interruptFreq = 0);
+	~Infogrames();
+
+	Instruments *getInstruments() const { return _instruments; }
+	bool getRepeating() const { return _repCount != 0; }
+	void setRepeating (int32 repCount) { _repCount = repCount; }
+
+	bool load(Common::SeekableReadStream &dum);
+	bool load(const char *dum);
+	void unload();
+	void restart() {
+		if (_data) {
+			// Use the mutex here to ensure we do not call init()
+			// while data is being read by the mixer thread.
+			_mutex.lock();
+			init();
+			startPlay();
+			_mutex.unlock();
+		}
+	}
+
+protected:
+	Instruments *_instruments;
+
+	static const uint8 tickCount[];
+	static const uint16 periods[];
+	byte *_data;
+	int32 _repCount;
+
+	byte *_subSong;
+	byte *_cmdBlocks;
+	byte *_volSlideBlocks;
+	byte *_periodSlideBlocks;
+	uint8 _speedCounter;
+	uint8 _speed;
+
+	uint16 _volume;
+	int16 _period;
+	uint8 _sample;
+
+	struct Slide {
+		byte *data;
+		int8 amount;
+		uint8 dataOffset;
+		int16 finetuneNeg;
+		int16 finetunePos;
+		uint8 curDelay1;
+		uint8 curDelay2;
+		uint8 flags; // 0: Apply finetune modifier, 2: Don't slide, 7: Continuous
+	};
+	struct Channel {
+		byte *cmdBlockIndices;
+		byte *cmdBlocks;
+		byte *cmds;
+		uint8 ticks;
+		uint8 tickCount;
+		Slide volSlide;
+		Slide periodSlide;
+		int16 period;
+		int8 periodMod;
+		uint8 flags; // 0: Need init, 5: Loop cmdBlocks, 6: Ignore channel
+	} _chn[4];
+
+	void init();
+	void reset();
+	void getNextSample(Channel &chn);
+	int16 tune(Slide &slide, int16 start) const;
+	virtual void interrupt();
+};
+
+} // End of namespace Audio
+
+#endif
diff --git a/audio/mods/maxtrax.cpp b/audio/mods/maxtrax.cpp
new file mode 100644
index 0000000000..a577c72eed
--- /dev/null
+++ b/audio/mods/maxtrax.cpp
@@ -0,0 +1,1040 @@
+/* ScummVM - Graphic Adventure Engine
+ *
+ * ScummVM is the legal property of its developers, whose names
+ * are too numerous to list here. Please refer to the COPYRIGHT
+ * file distributed with this source distribution.
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
+ *
+ * $URL$
+ * $Id$
+ *
+ */
+
+#include "common/scummsys.h"
+#include "common/endian.h"
+#include "common/stream.h"
+#include "common/util.h"
+#include "common/debug.h"
+
+#include "audio/mods/maxtrax.h"
+
+// test for engines using this class.
+#if defined(SOUND_MODS_MAXTRAX_H)
+
+namespace {
+
+enum { K_VALUE = 0x9fd77, PREF_PERIOD = 0x8fd77, PERIOD_LIMIT = 0x6f73d };
+enum { NO_BEND = 64 << 7, MAX_BEND_RANGE = 24 };
+
+int32 precalcNote(byte baseNote, int16 tune, byte octave) {
+	return K_VALUE + 0x3C000 - ((baseNote << 14) + (tune << 11) / 3) / 3 - (octave << 16);
+}
+
+int32 calcVolumeDelta(int32 delta, uint16 time, uint16 vBlankFreq) {
+	const int32 div = time * vBlankFreq;
+	// div <= 1000 means time to small (or even 0)
+	return (div <= 1000) ? delta : (1000 * delta) / div;
+}
+
+int32 calcTempo(const uint16 tempo, uint16 vBlankFreq) {
+	return (int32)(((uint32)(tempo & 0xFFF0) << 8) / (uint16)(5 * vBlankFreq));
+}
+
+void nullFunc(int) {}
+
+// Function to calculate 2^x, where x is a fixedpoint number with 16 fraction bits
+// using exp would be more accurate and needs less space if mathlibrary is already linked
+// but this function should be faster and doesnt use floats
+#if 1
+inline uint32 pow2Fixed(int32 val) {
+	static const uint16 tablePow2[] = {
+			0,   178,   356,   535,   714,   893,  1073,  1254,  1435,  1617,  1799,  1981,  2164,  2348,  2532,  2716,
+		 2902,  3087,  3273,  3460,  3647,  3834,  4022,  4211,  4400,  4590,  4780,  4971,  5162,  5353,  5546,  5738,
+		 5932,  6125,  6320,  6514,  6710,  6906,  7102,  7299,  7496,  7694,  7893,  8092,  8292,  8492,  8693,  8894,
+		 9096,  9298,  9501,  9704,  9908, 10113, 10318, 10524, 10730, 10937, 11144, 11352, 11560, 11769, 11979, 12189,
+		12400, 12611, 12823, 13036, 13249, 13462, 13676, 13891, 14106, 14322, 14539, 14756, 14974, 15192, 15411, 15630,
+		15850, 16071, 16292, 16514, 16737, 16960, 17183, 17408, 17633, 17858, 18084, 18311, 18538, 18766, 18995, 19224,
+		19454, 19684, 19915, 20147, 20379, 20612, 20846, 21080, 21315, 21550, 21786, 22023, 22260, 22498, 22737, 22977,
+		23216, 23457, 23698, 23940, 24183, 24426, 24670, 24915, 25160, 25406, 25652, 25900, 26148, 26396, 26645, 26895,
+		27146, 27397, 27649, 27902, 28155, 28409, 28664, 28919, 29175, 29432, 29690, 29948, 30207, 30466, 30727, 30988,
+		31249, 31512, 31775, 32039, 32303, 32568, 32834, 33101, 33369, 33637, 33906, 34175, 34446, 34717, 34988, 35261,
+		35534, 35808, 36083, 36359, 36635, 36912, 37190, 37468, 37747, 38028, 38308, 38590, 38872, 39155, 39439, 39724,
+		40009, 40295, 40582, 40870, 41158, 41448, 41738, 42029, 42320, 42613, 42906, 43200, 43495, 43790, 44087, 44384,
+		44682, 44981, 45280, 45581, 45882, 46184, 46487, 46791, 47095, 47401, 47707, 48014, 48322, 48631, 48940, 49251,
+		49562, 49874, 50187, 50500, 50815, 51131, 51447, 51764, 52082, 52401, 52721, 53041, 53363, 53685, 54008, 54333,
+		54658, 54983, 55310, 55638, 55966, 56296, 56626, 56957, 57289, 57622, 57956, 58291, 58627, 58964, 59301, 59640,
+		59979, 60319, 60661, 61003, 61346, 61690, 62035, 62381, 62727, 63075, 63424, 63774, 64124, 64476, 64828, 65182,
+			0
+	};
+	const uint16 whole = val >> 16;
+	const uint8 index = (uint8)(val >> 8);
+	// calculate fractional part.
+	const uint16 base = tablePow2[index];
+	// linear interpolation and add 1.0
+	uint32 exponent = ((uint32)(uint16)(tablePow2[index + 1] - base) * (uint8)val) + ((uint32)base << 8) + (1 << 24);
+
+	if (whole < 24) {
+		// shift away all but the last fractional bit which is used for rounding,
+		// then round to nearest integer
+		exponent = ((exponent >> (23 - whole)) + 1) >> 1;
+	} else if (whole < 32) {
+		// no need to round here
+		exponent <<= whole - 24;
+	} else if (val > 0) {
+		// overflow
+		exponent = 0xFFFFFFFF;
+	} else  {
+		// negative integer, test if >= -0.5
+		exponent = (val >= -0x8000) ? 1 : 0;
+	}
+	return exponent;
+}
+#else
+inline uint32 pow2Fixed(int32 val) {
+	return (uint32)(expf((float)val * (float)(0.69314718055994530942 / (1 << 16))) + 0.5f);
+}
+#endif
+
+}	// End of namespace
+
+namespace Audio {
+
+MaxTrax::MaxTrax(int rate, bool stereo, uint16 vBlankFreq, uint16 maxScores)
+	: Paula(stereo, rate, rate / vBlankFreq),
+	  _patch(),
+	  _scores(),
+	  _numScores() {
+	_playerCtx.maxScoreNum = maxScores;
+	_playerCtx.vBlankFreq = vBlankFreq;
+	_playerCtx.frameUnit = (uint16)((1000 << 8) /  vBlankFreq);
+	_playerCtx.scoreIndex = -1;
+	_playerCtx.volume = 0x40;
+
+	_playerCtx.tempo = 120;
+	_playerCtx.tempoTime = 0;
+	_playerCtx.filterOn = true;
+	_playerCtx.syncCallBack = &nullFunc;
+
+	resetPlayer();
+	for (int i = 0; i < ARRAYSIZE(_channelCtx); ++i)
+		_channelCtx[i].regParamNumber = 0;
+}
+
+MaxTrax::~MaxTrax() {
+	stopMusic();
+	freePatches();
+	freeScores();
+}
+
+void MaxTrax::interrupt() {
+	// a5 - maxtraxm a4 . globaldata
+
+	// TODO
+	// test for changes in shared struct and make changes
+	// specifically all used channels get marked altered
+
+	_playerCtx.ticks += _playerCtx.tickUnit;
+	const int32 millis = _playerCtx.ticks >> 8; // d4
+
+	for (int i = 0; i < ARRAYSIZE(_voiceCtx); ++i) {
+		VoiceContext &voice = _voiceCtx[i];
+		if (voice.stopEventTime >= 0) {
+			assert(voice.channel);
+			voice.stopEventTime -= (voice.channel < &_channelCtx[kNumChannels]) ? _playerCtx.tickUnit : _playerCtx.frameUnit;
+			if (voice.stopEventTime <= 0 && voice.status > VoiceContext::kStatusRelease) {
+				if ((voice.channel->flags & ChannelContext::kFlagDamper) != 0)
+					voice.hasDamper = true;
+				else
+					voice.status = VoiceContext::kStatusRelease;
+			}
+		}
+	}
+
+	if (_playerCtx.scoreIndex >= 0) {
+		const Event *curEvent = _playerCtx.nextEvent;
+		int32 eventDelta = _playerCtx.nextEventTime - millis;
+		for (; eventDelta <= 0; eventDelta += (++curEvent)->startTime) {
+			const byte cmd = curEvent->command;
+			ChannelContext &channel = _channelCtx[curEvent->parameter & 0x0F];
+
+			// outPutEvent(*curEvent);
+			// debug("CurTime, EventDelta, NextDelta: %d, %d, %d", millis, eventDelta, eventDelta + curEvent[1].startTime );
+
+			if (cmd < 0x80) {	// Note
+				const int8 voiceIndex = noteOn(channel, cmd, (curEvent->parameter & 0xF0) >> 1, kPriorityScore);
+				if (voiceIndex >= 0)
+					_voiceCtx[voiceIndex].stopEventTime = MAX<int32>(0, (eventDelta + curEvent->stopTime) << 8);
+
+			} else {
+				switch (cmd) {
+
+				case 0x80:	// TEMPO
+					if ((_playerCtx.tickUnit >> 8) > curEvent->stopTime) {
+						_playerCtx.tickUnit = calcTempo(curEvent->parameter << 4, _playerCtx.vBlankFreq);
+						_playerCtx.tempoTime = 0;
+					} else {
+						_playerCtx.tempoStart = _playerCtx.tempo;
+						_playerCtx.tempoDelta = (curEvent->parameter << 4) - _playerCtx.tempoStart;
+						_playerCtx.tempoTime  = (curEvent->stopTime << 8);
+						_playerCtx.tempoTicks = 0;
+					}
+					break;
+
+				case 0xC0:	// PROGRAM
+					channel.patch = &_patch[curEvent->stopTime & (kNumPatches - 1)];
+					break;
+
+				case 0xE0:	// BEND
+					channel.pitchBend = ((curEvent->stopTime & 0x7F00) >> 1) | (curEvent->stopTime & 0x7f);
+					channel.pitchReal = (((int32)channel.pitchBendRange * channel.pitchBend) >> 5) - (channel.pitchBendRange << 8);
+					channel.isAltered = true;
+					break;
+
+				case 0xFF:	// END
+					if (_playerCtx.musicLoop) {
+						curEvent = _scores[_playerCtx.scoreIndex].events;
+						eventDelta = curEvent->startTime - millis;
+						_playerCtx.ticks = 0;
+					} else
+						_playerCtx.scoreIndex = -1;
+					// stop processing for this tick
+					goto endOfEventLoop;
+
+				case 0xA0: 	// SPECIAL
+					switch (curEvent->stopTime >> 8){
+					case 0x01:	// SPECIAL_SYNC
+						_playerCtx.syncCallBack(curEvent->stopTime & 0xFF);
+						break;
+					case 0x02:	// SPECIAL_BEGINREP
+						// we allow a depth of 4 loops
+						for (int i = 0; i < ARRAYSIZE(_playerCtx.repeatPoint); ++i) {
+							if (!_playerCtx.repeatPoint[i]) {
+								_playerCtx.repeatPoint[i] = curEvent;
+								_playerCtx.repeatCount[i] = curEvent->stopTime & 0xFF;
+								break;
+							}
+						}
+						break;
+					case 0x03:	// SPECIAL_ENDREP
+						for (int i = ARRAYSIZE(_playerCtx.repeatPoint) - 1; i >= 0; --i) {
+							if (_playerCtx.repeatPoint[i]) {
+								if (_playerCtx.repeatCount[i]--)
+									curEvent = _playerCtx.repeatPoint[i]; // gets incremented by 1 at end of loop
+								else
+									_playerCtx.repeatPoint[i] = 0;
+								break;
+							}
+						}
+						break;
+					}
+					break;
+
+				case 0xB0:	// CONTROL
+					controlCh(channel, (byte)(curEvent->stopTime >> 8), (byte)curEvent->stopTime);
+					break;
+
+				default:
+					debug("Unhandled Command");
+					outPutEvent(*curEvent);
+				}
+			}
+		}
+endOfEventLoop:
+		_playerCtx.nextEvent = curEvent;
+		_playerCtx.nextEventTime = eventDelta + millis;
+
+		// tempoEffect
+		if (_playerCtx.tempoTime) {
+			_playerCtx.tempoTicks += _playerCtx.tickUnit;
+			uint16 newTempo = _playerCtx.tempoStart;
+			if (_playerCtx.tempoTicks < _playerCtx.tempoTime) {
+				newTempo += (uint16)((_playerCtx.tempoTicks * _playerCtx.tempoDelta) / _playerCtx.tempoTime);
+			} else {
+				_playerCtx.tempoTime = 0;
+				newTempo += _playerCtx.tempoDelta;
+			}
+			_playerCtx.tickUnit = calcTempo(newTempo, _playerCtx.vBlankFreq);
+		}
+	}
+
+	// Handling of Envelopes and Portamento
+	for (int i = 0; i < ARRAYSIZE(_voiceCtx); ++i) {
+		VoiceContext &voice = _voiceCtx[i];
+		if (!voice.channel)
+			continue;
+		const ChannelContext &channel = *voice.channel;
+		const Patch &patch = *voice.patch;
+
+		switch (voice.status) {
+		case VoiceContext::kStatusSustain:
+			// we need to check if some voices have no sustainSample.
+			// in that case they are finished after the attackSample is done
+			if (voice.dmaOff && Paula::getChannelDmaCount((byte)i) >= voice.dmaOff ) {
+				voice.dmaOff = 0;
+				voice.isBlocked = 0;
+				voice.priority = 0;
+				// disable it in next tick
+				voice.stopEventTime = 0;
+			}
+			if (!channel.isAltered && !voice.hasPortamento && !channel.modulation)
+				continue;
+			// Update Volume and Period
+			break;
+
+		case VoiceContext::kStatusHalt:
+			killVoice((byte)i);
+			continue;
+
+		case VoiceContext::kStatusStart:
+			if (patch.attackLen) {
+				voice.envelope = patch.attackPtr;
+				const uint16 duration = voice.envelope->duration;
+				voice.envelopeLeft = patch.attackLen;
+				voice.ticksLeft = duration << 8;
+				voice.status = VoiceContext::kStatusAttack;
+				voice.incrVolume = calcVolumeDelta((int32)voice.envelope->volume, duration, _playerCtx.vBlankFreq);
+				// Process Envelope
+			} else {
+				voice.status = VoiceContext::kStatusSustain;
+				voice.baseVolume = patch.volume;
+				// Update Volume and Period
+			}
+			break;
+
+		case VoiceContext::kStatusRelease:
+			if (patch.releaseLen) {
+				voice.envelope = patch.attackPtr + patch.attackLen;
+				const uint16 duration = voice.envelope->duration;
+				voice.envelopeLeft = patch.releaseLen;
+				voice.ticksLeft = duration << 8;
+				voice.status = VoiceContext::kStatusDecay;
+				voice.incrVolume = calcVolumeDelta((int32)voice.envelope->volume - voice.baseVolume, duration, _playerCtx.vBlankFreq);
+				// Process Envelope
+			} else {
+				voice.status = VoiceContext::kStatusHalt;
+				voice.lastVolume = 0;
+				// Send Audio Packet
+			}
+			voice.stopEventTime = -1;
+			break;
+		}
+
+		// Process Envelope
+		const uint16 envUnit = _playerCtx.frameUnit;
+		if (voice.envelope) {
+			if (voice.ticksLeft > envUnit) {	// envelope still active
+				voice.baseVolume = (uint16) MIN<int32>(MAX<int32>(0, voice.baseVolume + voice.incrVolume), 0x8000);
+				voice.ticksLeft -= envUnit;
+				// Update Volume and Period
+
+			} else {	// next or last Envelope
+				voice.baseVolume = voice.envelope->volume;
+				assert(voice.envelopeLeft > 0);
+				if (--voice.envelopeLeft) {
+					++voice.envelope;
+					const uint16 duration = voice.envelope->duration;
+					voice.ticksLeft = duration << 8;
+					voice.incrVolume = calcVolumeDelta((int32)voice.envelope->volume - voice.baseVolume, duration, _playerCtx.vBlankFreq);
+					// Update Volume and Period
+				} else if (voice.status == VoiceContext::kStatusDecay) {
+					voice.status = VoiceContext::kStatusHalt;
+					voice.envelope = 0;
+					voice.lastVolume = 0;
+					// Send Audio Packet
+				} else {
+					assert(voice.status == VoiceContext::kStatusAttack);
+					voice.status = VoiceContext::kStatusSustain;
+					voice.envelope = 0;
+					// Update Volume and Period
+				}
+			}
+		}
+
+		// Update Volume and Period
+		if (voice.status >= VoiceContext::kStatusDecay) {
+			// Calc volume
+			uint16 vol = (voice.noteVolume < (1 << 7)) ? (voice.noteVolume * _playerCtx.volume) >> 7 : _playerCtx.volume;
+			if (voice.baseVolume < (1 << 15))
+				vol = (uint16)(((uint32)vol * voice.baseVolume) >> 15);
+			if (voice.channel->volume < (1 << 7))
+				vol = (vol * voice.channel->volume) >> 7;
+			voice.lastVolume = (byte)MIN(vol, (uint16)0x64);
+
+			// Calc Period
+			if (voice.hasPortamento) {
+				voice.portaTicks += envUnit;
+				if ((uint16)(voice.portaTicks >> 8) >= channel.portamentoTime) {
+					voice.hasPortamento = false;
+					voice.baseNote = voice.endNote;
+					voice.preCalcNote = precalcNote(voice.baseNote, patch.tune, voice.octave);
+				}
+				voice.lastPeriod = calcNote(voice);
+			} else if (channel.isAltered || channel.modulation)
+				voice.lastPeriod = calcNote(voice);
+		}
+
+		// Send Audio Packet
+		Paula::setChannelPeriod((byte)i, (voice.lastPeriod) ? voice.lastPeriod : 1000);
+		Paula::setChannelVolume((byte)i, (voice.lastPeriod) ? voice.lastVolume : 0);
+	}
+	for (ChannelContext *c = _channelCtx; c != &_channelCtx[ARRAYSIZE(_channelCtx)]; ++c)
+		c->isAltered = false;
+
+#ifdef MAXTRAX_HAS_MODULATION
+	// original player had _playerCtx.sineValue = _playerCtx.frameUnit >> 2
+	// this should fit the comments that modtime=1000 is one second ?
+	_playerCtx.sineValue += _playerCtx.frameUnit;
+#endif
+}
+
+void MaxTrax::controlCh(ChannelContext &channel, const byte command, const byte data) {
+	switch (command) {
+	case 0x01:	// modulation level MSB
+		channel.modulation = data << 8;
+		break;
+	case 0x21:	// modulation level LSB
+		channel.modulation = (channel.modulation & 0xFF00) || ((data * 2) & 0xFF);
+		break;
+	case 0x05:	// portamento time MSB
+		channel.portamentoTime = data << 7;
+		break;
+	case 0x25:	// portamento time LSB
+		channel.portamentoTime = (channel.portamentoTime & 0x3f80) || data;
+		break;
+	case 0x06:	// data entry MSB
+		if (channel.regParamNumber == 0) {
+			channel.pitchBendRange = (int8)MIN((uint8)MAX_BEND_RANGE, (uint8)data);
+			channel.pitchReal = (((int32)channel.pitchBendRange * channel.pitchBend) >> 5) - (channel.pitchBendRange << 8);
+			channel.isAltered = true;
+		}
+		break;
+	case 0x07:	// Main Volume MSB
+		channel.volume = (data == 0) ? 0 : data + 1;
+		channel.isAltered = true;
+		break;
+	case 0x0A:	// Pan
+		if (data > 0x40 || (data == 0x40 && ((&channel - _channelCtx) & 1) != 0))
+			channel.flags |= ChannelContext::kFlagRightChannel;
+		else
+			channel.flags &= ~ChannelContext::kFlagRightChannel;
+		break;
+	case 0x10:	// GPC as Modulation Time MSB
+		channel.modulationTime = data << 7;
+		break;
+	case 0x30:	// GPC as Modulation Time LSB
+		channel.modulationTime = (channel.modulationTime & 0x3f80) || data;
+		break;
+	case 0x11:	// GPC as Microtonal Set MSB
+		channel.microtonal = data << 8;
+		break;
+	case 0x31:	// GPC as Microtonal Set LSB
+		channel.microtonal = (channel.microtonal & 0xFF00) || ((data * 2) & 0xFF);
+		break;
+	case 0x40:	// Damper Pedal
+		if ((data & 0x40) != 0)
+			channel.flags |= ChannelContext::kFlagDamper;
+		else {
+			channel.flags &= ~ChannelContext::kFlagDamper;
+			// release all dampered voices on this channel
+			for (int i = 0; i < ARRAYSIZE(_voiceCtx); ++i) {
+				if (_voiceCtx[i].channel == &channel && _voiceCtx[i].hasDamper) {
+					_voiceCtx[i].hasDamper = false;
+					_voiceCtx[i].status = VoiceContext::kStatusRelease;
+				}
+			}
+		}
+		break;
+	case 0x41:	// Portamento off/on
+		if ((data & 0x40) != 0)
+			channel.flags |= ChannelContext::kFlagPortamento;
+		else
+			channel.flags &= ~ChannelContext::kFlagPortamento;
+		break;
+	case 0x50:	// Microtonal off/on
+		if ((data & 0x40) != 0)
+			channel.flags |= ChannelContext::kFlagMicrotonal;
+		else
+			channel.flags &= ~ChannelContext::kFlagMicrotonal;
+		break;
+	case 0x51:	// Audio Filter off/on
+		Paula::setAudioFilter(data > 0x40 || (data == 0x40 && _playerCtx.filterOn));
+		break;
+	case 0x65:	// RPN MSB
+		channel.regParamNumber = (data << 8) || (channel.regParamNumber & 0xFF);
+		break;
+	case 0x64:	// RPN LSB
+		channel.regParamNumber = (channel.regParamNumber & 0xFF00) || data;
+		break;
+	case 0x79:	// Reset All Controllers
+		resetChannel(channel, ((&channel - _channelCtx) & 1) != 0);
+		break;
+	case 0x7E:	// MONO mode
+		channel.flags |= ChannelContext::kFlagMono;
+		goto allNotesOff;
+	case 0x7F:	// POLY mode
+		channel.flags &= ~ChannelContext::kFlagMono;
+		// Fallthrough
+	case 0x7B:	// All Notes Off
+allNotesOff:
+		for (int i = 0; i < ARRAYSIZE(_voiceCtx); ++i) {
+			if (_voiceCtx[i].channel == &channel) {
+				if ((channel.flags & ChannelContext::kFlagDamper) != 0)
+					_voiceCtx[i].hasDamper = true;
+				else
+					_voiceCtx[i].status = VoiceContext::kStatusRelease;
+			}
+		}
+		break;
+	case 0x78:	// All Sounds Off
+		for (int i = 0; i < ARRAYSIZE(_voiceCtx); ++i) {
+			if (_voiceCtx[i].channel == &channel)
+				killVoice((byte)i);
+		}
+		break;
+	}
+}
+
+void MaxTrax::setTempo(const uint16 tempo) {
+	Common::StackLock lock(_mutex);
+	_playerCtx.tickUnit = calcTempo(tempo, _playerCtx.vBlankFreq);
+}
+
+void MaxTrax::resetPlayer() {
+	for (int i = 0; i < ARRAYSIZE(_voiceCtx); ++i)
+		killVoice((byte)i);
+
+	for (int i = 0; i < ARRAYSIZE(_channelCtx); ++i) {
+		_channelCtx[i].flags = 0;
+		_channelCtx[i].lastNote = (uint8)-1;
+		resetChannel(_channelCtx[i], (i & 1) != 0);
+		_channelCtx[i].patch = (i < kNumChannels) ? &_patch[i] : 0;
+	}
+
+#ifdef MAXTRAX_HAS_MICROTONAL
+	for (int i = 0; i < ARRAYSIZE(_microtonal); ++i)
+		_microtonal[i] = (int16)(i << 8);
+#endif
+}
+
+void MaxTrax::stopMusic() {
+	Common::StackLock lock(_mutex);
+	_playerCtx.scoreIndex = -1;
+	for (int i = 0; i < ARRAYSIZE(_voiceCtx); ++i) {
+		if (_voiceCtx[i].channel < &_channelCtx[kNumChannels])
+			killVoice((byte)i);
+	}
+}
+
+bool MaxTrax::playSong(int songIndex, bool loop) {
+	if (songIndex < 0 || songIndex >= _numScores)
+		return false;
+	Common::StackLock lock(_mutex);
+	_playerCtx.scoreIndex = -1;
+	resetPlayer();
+	for (int i = 0; i < ARRAYSIZE(_playerCtx.repeatPoint); ++i)
+		_playerCtx.repeatPoint[i] = 0;
+
+	setTempo(_playerCtx.tempoInitial << 4);
+	Paula::setAudioFilter(_playerCtx.filterOn);
+	_playerCtx.musicLoop = loop;
+	_playerCtx.tempoTime = 0;
+	_playerCtx.scoreIndex = songIndex;
+	_playerCtx.ticks = 0;
+
+	_playerCtx.nextEvent = _scores[songIndex].events;
+	_playerCtx.nextEventTime = _playerCtx.nextEvent->startTime;
+
+	Paula::startPaula();
+	return true;
+}
+
+void MaxTrax::advanceSong(int advance) {
+	Common::StackLock lock(_mutex);
+	if (_playerCtx.scoreIndex >= 0) {
+		const Event *cev = _playerCtx.nextEvent;
+		if (cev) {
+			for (; advance > 0; --advance) {
+				// TODO - check for boundaries
+				for (; cev->command != 0xFF && (cev->command != 0xA0 || (cev->stopTime >> 8) != 0x00); ++cev)
+					; // no end_command or special_command + end
+			}
+			_playerCtx.nextEvent = cev;
+		}
+	}
+}
+
+void MaxTrax::killVoice(byte num) {
+	VoiceContext &voice = _voiceCtx[num];
+	voice.channel = 0;
+	voice.envelope = 0;
+	voice.status = VoiceContext::kStatusFree;
+	voice.isBlocked = 0;
+	voice.hasDamper = false;
+	voice.hasPortamento = false;
+	voice.priority = 0;
+	voice.stopEventTime = -1;
+	voice.dmaOff = 0;
+	voice.lastVolume = 0;
+	voice.tieBreak = 0;
+	//voice.uinqueId = 0;
+
+	// "stop" voice, set period to 1, vol to 0
+	Paula::disableChannel(num);
+	Paula::setChannelPeriod(num, 1);
+	Paula::setChannelVolume(num, 0);
+}
+
+int8 MaxTrax::pickvoice(uint pick, int16 pri) {
+	enum { kPrioFlagFixedSide = 1 << 3 };
+	pick &= 3;
+	if ((pri & (kPrioFlagFixedSide)) == 0) {
+		const bool leftSide = (uint)(pick - 1) > 1;
+		const int leftBest = MIN(_voiceCtx[0].status, _voiceCtx[3].status);
+		const int rightBest = MIN(_voiceCtx[1].status, _voiceCtx[2].status);
+		const int sameSide = (leftSide) ? leftBest : rightBest;
+		const int otherSide = leftBest + rightBest - sameSide;
+
+		if (sameSide > VoiceContext::kStatusRelease && otherSide <= VoiceContext::kStatusRelease)
+			pick ^= 1; // switches sides
+	}
+	pri &= ~kPrioFlagFixedSide;
+
+	for (int i = 2; i > 0; --i) {
+		VoiceContext *voice = &_voiceCtx[pick];
+		VoiceContext *alternate = &_voiceCtx[pick ^ 3];
+
+		const uint16 voiceVal = voice->status << 8 | voice->lastVolume;
+		const uint16 altVal = alternate->status << 8 | alternate->lastVolume;
+
+		if (voiceVal + voice->tieBreak > altVal
+			|| voice->isBlocked > alternate->isBlocked) {
+
+			// this is somewhat different to the original player,
+			// but has a similar result
+			voice->tieBreak = 0;
+			alternate->tieBreak = 1;
+
+			pick ^= 3; // switch channels
+			VoiceContext *tmp = voice;
+			voice = alternate;
+			alternate = tmp;
+		}
+
+		if (voice->isBlocked || voice->priority > pri) {
+			// if not already done, switch sides and try again
+			pick ^= 1;
+			continue;
+		}
+		// succeded
+		return (int8)pick;
+	}
+	// failed
+	debug(5, "MaxTrax: could not find channel for note");
+	return -1;
+}
+
+uint16 MaxTrax::calcNote(const VoiceContext &voice) {
+	const ChannelContext &channel = *voice.channel;
+	int16 bend = channel.pitchReal;
+
+#ifdef MAXTRAX_HAS_MICROTONAL
+	if (voice.hasPortamento) {
+		if ((channel.flags & ChannelContext::kFlagMicrotonal) != 0)
+			bend += (int16)(((_microtonal[voice.endNote] - _microtonal[voice.baseNote]) * voice.portaTicks) >> 8) / channel.portamentoTime;
+		else
+			bend += (int16)(((int8)(voice.endNote - voice.baseNote)) * voice.portaTicks) / channel.portamentoTime;
+	}
+
+	if ((channel.flags & ChannelContext::kFlagMicrotonal) != 0)
+		bend += _microtonal[voice.baseNote];
+#else
+	if (voice.hasPortamento)
+		bend += (int16)(((int8)(voice.endNote - voice.baseNote)) * voice.portaTicks) / channel.portamentoTime;
+#endif
+
+#ifdef MAXTRAX_HAS_MODULATION
+	static const uint8 tableSine[] = {
+		  0,   5,  12,  18,  24,  30,  37,  43,  49,  55,  61,  67,  73,  79,  85,  91,
+		 97, 103, 108, 114, 120, 125, 131, 136, 141, 146, 151, 156, 161, 166, 171, 176,
+		180, 184, 189, 193, 197, 201, 205, 208, 212, 215, 219, 222, 225, 228, 230, 233,
+		236, 238, 240, 242, 244, 246, 247, 249, 250, 251, 252, 253, 254, 254, 255, 255,
+		255, 255, 255, 254, 254, 253, 252, 251, 250, 249, 247, 246, 244, 242, 240, 238,
+		236, 233, 230, 228, 225, 222, 219, 215, 212, 208, 205, 201, 197, 193, 189, 184,
+		180, 176, 171, 166, 161, 156, 151, 146, 141, 136, 131, 125, 120, 114, 108, 103,
+		 97,  91,  85,  79,  73,  67,  61,  55,  49,  43,  37,  30,  24,  18,  12,   5
+	};
+	if (channel.modulation) {
+		if ((channel.flags & ChannelContext::kFlagModVolume) == 0) {
+			const uint8 sineByte = _playerCtx.sineValue / channel.modulationTime;
+			const uint8 sineIndex = sineByte & 0x7F;
+			const int16 modVal = ((uint32)(uint16)(tableSine[sineIndex] + (sineIndex ? 1 : 0)) * channel.modulation) >> 8;
+			bend = (sineByte < 0x80) ? bend + modVal : bend - modVal;
+		}
+	}
+#endif
+
+	// tone = voice.baseNote << 8 + microtonal
+	// bend = channelPitch + porta + modulation
+
+	const int32 tone = voice.preCalcNote + (bend << 6) / 3;
+
+	return (tone >= PERIOD_LIMIT) ? (uint16)pow2Fixed(tone) : 0;
+}
+
+int8 MaxTrax::noteOn(ChannelContext &channel, const byte note, uint16 volume, uint16 pri) {
+#ifdef MAXTRAX_HAS_MICROTONAL
+	if (channel.microtonal >= 0)
+		_microtonal[note % 127] = channel.microtonal;
+#endif
+
+	if (!volume)
+		return -1;
+
+	const Patch &patch = *channel.patch;
+	if (!patch.samplePtr || patch.sampleTotalLen == 0)
+		return -1;
+	int8 voiceNum = -1;
+	if ((channel.flags & ChannelContext::kFlagMono) == 0) {
+		voiceNum = pickvoice((channel.flags & ChannelContext::kFlagRightChannel) != 0 ? 1 : 0, pri);
+	} else {
+		VoiceContext *voice = _voiceCtx + ARRAYSIZE(_voiceCtx) - 1;
+		for (voiceNum = ARRAYSIZE(_voiceCtx) - 1; voiceNum >= 0 && voice->channel != &channel; --voiceNum, --voice)
+			;
+		if (voiceNum < 0)
+			voiceNum = pickvoice((channel.flags & ChannelContext::kFlagRightChannel) != 0 ? 1 : 0, pri);
+		else if (voice->status >= VoiceContext::kStatusSustain && (channel.flags & ChannelContext::kFlagPortamento) != 0) {
+			// reset previous porta
+			if (voice->hasPortamento)
+				voice->baseNote = voice->endNote;
+			voice->preCalcNote = precalcNote(voice->baseNote, patch.tune, voice->octave);
+			voice->noteVolume = (_playerCtx.handleVolume) ? volume + 1 : 128;
+			voice->portaTicks = 0;
+			voice->hasPortamento = true;
+			voice->endNote = channel.lastNote = note;
+			return voiceNum;
+		}
+	}
+
+	if (voiceNum >= 0) {
+		VoiceContext &voice = _voiceCtx[voiceNum];
+		voice.hasDamper = false;
+		voice.isBlocked = 0;
+		voice.hasPortamento = false;
+		if (voice.channel)
+			killVoice(voiceNum);
+		voice.channel = &channel;
+		voice.patch = &patch;
+		voice.baseNote = note;
+
+		// always base octave on the note in the command, regardless of porta
+		const int32 plainNote = precalcNote(note, patch.tune, 0);
+		// calculate which sample to use
+		const int useOctave = (plainNote <= PREF_PERIOD) ? 0 : MIN<int32>((plainNote + 0xFFFF - PREF_PERIOD) >> 16, patch.sampleOctaves - 1);
+		voice.octave = (byte)useOctave;
+		// adjust precalculated value
+		voice.preCalcNote = plainNote - (useOctave << 16);
+
+		// next calculate the actual period which depends on wether porta is enabled
+		if (&channel < &_channelCtx[kNumChannels] && (channel.flags & ChannelContext::kFlagPortamento) != 0) {
+			if ((channel.flags & ChannelContext::kFlagMono) != 0 && channel.lastNote < 0x80 && channel.lastNote != note) {
+				voice.portaTicks = 0;
+				voice.baseNote = channel.lastNote;
+				voice.endNote = note;
+				voice.hasPortamento = true;
+				voice.preCalcNote = precalcNote(voice.baseNote, patch.tune, voice.octave);
+			}
+			channel.lastNote = note;
+		}
+
+		voice.lastPeriod = calcNote(voice);
+
+		voice.priority = (byte)pri;
+		voice.status = VoiceContext::kStatusStart;
+
+		voice.noteVolume = (_playerCtx.handleVolume) ? volume + 1 : 128;
+		voice.baseVolume = 0;
+
+		// TODO: since the original player is using the OS-functions, more than 1 sample could be queued up already
+		// get samplestart for the given octave
+		const int8 *samplePtr = patch.samplePtr + (patch.sampleTotalLen << useOctave) - patch.sampleTotalLen;
+		if (patch.sampleAttackLen) {
+			Paula::setChannelSampleStart(voiceNum, samplePtr);
+			Paula::setChannelSampleLen(voiceNum, (patch.sampleAttackLen << useOctave) / 2);
+
+			Paula::enableChannel(voiceNum);
+			// wait  for dma-clear
+		}
+
+		if (patch.sampleTotalLen > patch.sampleAttackLen) {
+			Paula::setChannelSampleStart(voiceNum, samplePtr + (patch.sampleAttackLen << useOctave));
+			Paula::setChannelSampleLen(voiceNum, ((patch.sampleTotalLen - patch.sampleAttackLen) << useOctave) / 2);
+			if (!patch.sampleAttackLen)
+				Paula::enableChannel(voiceNum); // need to enable channel
+			// another pointless wait for DMA-Clear???
+
+		} else { // no sustain sample
+			// this means we must stop playback after the attacksample finished
+			// so we queue up an "empty" sample and note that we need to kill the sample after dma finished
+			Paula::setChannelSampleStart(voiceNum, 0);
+			Paula::setChannelSampleLen(voiceNum, 0);
+			Paula::setChannelDmaCount(voiceNum);
+			voice.dmaOff = 1;
+		}
+
+		Paula::setChannelPeriod(voiceNum, (voice.lastPeriod) ? voice.lastPeriod : 1000);
+		Paula::setChannelVolume(voiceNum, 0);
+	}
+	return voiceNum;
+}
+
+void MaxTrax::resetChannel(ChannelContext &chan, bool rightChannel) {
+	chan.modulation = 0;
+	chan.modulationTime = 1000;
+	chan.microtonal = -1;
+	chan.portamentoTime = 500;
+	chan.pitchBend = NO_BEND;
+	chan.pitchReal = 0;
+	chan.pitchBendRange = MAX_BEND_RANGE;
+	chan.volume = 128;
+	chan.flags &= ~(ChannelContext::kFlagPortamento | ChannelContext::kFlagMicrotonal | ChannelContext::kFlagRightChannel);
+	chan.isAltered = true;
+	if (rightChannel)
+		chan.flags |= ChannelContext::kFlagRightChannel;
+}
+
+void MaxTrax::freeScores() {
+	if (_scores) {
+		for (int i = 0; i < _numScores; ++i)
+			delete[] _scores[i].events;
+		delete[] _scores;
+		_scores = 0;
+	}
+	_numScores = 0;
+	_playerCtx.tempo = 120;
+	_playerCtx.filterOn = true;
+}
+
+void MaxTrax::freePatches() {
+	for (int i = 0; i < ARRAYSIZE(_patch); ++i) {
+		delete[] _patch[i].samplePtr;
+		delete[] _patch[i].attackPtr;
+	}
+	memset(_patch, 0, sizeof(_patch));
+}
+
+void MaxTrax::setSignalCallback(void (*callback) (int)) {
+	Common::StackLock lock(_mutex);
+	_playerCtx.syncCallBack = (callback == 0) ? nullFunc : callback;
+}
+
+int MaxTrax::playNote(byte note, byte patch, uint16 duration, uint16 volume, bool rightSide) {
+	Common::StackLock lock(_mutex);
+	assert(patch < ARRAYSIZE(_patch));
+
+	ChannelContext &channel = _channelCtx[kNumChannels];
+	channel.flags = (rightSide) ? ChannelContext::kFlagRightChannel : 0;
+	channel.isAltered = false;
+	channel.patch = &_patch[patch];
+	const int8 voiceIndex = noteOn(channel, note, (byte)volume, kPriorityNote);
+	if (voiceIndex >= 0) {
+		_voiceCtx[voiceIndex].stopEventTime = duration << 8;
+		Paula::startPaula();
+	}
+	return voiceIndex;
+}
+
+bool MaxTrax::load(Common::SeekableReadStream &musicData, bool loadScores, bool loadSamples) {
+	Common::StackLock lock(_mutex);
+	stopMusic();
+	if (loadSamples)
+		freePatches();
+	if (loadScores)
+		freeScores();
+	const char *errorMsg = 0;
+	// 0x0000: 4 Bytes Header "MXTX"
+	// 0x0004: uint16 tempo
+	// 0x0006: uint16 flags. bit0 = lowpassfilter, bit1 = attackvolume, bit15 = microtonal
+	if (musicData.size() < 10 || musicData.readUint32BE() != 0x4D585458) {
+		warning("Maxtrax: File is not a Maxtrax Module");
+		return false;
+	}
+	const uint16 songTempo = musicData.readUint16BE();
+	const uint16 flags = musicData.readUint16BE();
+	if (loadScores) {
+		_playerCtx.tempoInitial = songTempo;
+		_playerCtx.filterOn = (flags & 1) != 0;
+		_playerCtx.handleVolume = (flags & 2) != 0;
+	}
+
+	if (flags & (1 << 15)) {
+		debug(5, "Maxtrax: Song has microtonal");
+#ifdef MAXTRAX_HAS_MICROTONAL
+		if (loadScores) {
+			for (int i = 0; i < ARRAYSIZE(_microtonal); ++i)
+				_microtonal[i] = musicData.readUint16BE();
+		} else
+			musicData.skip(128 * 2);
+#else
+		musicData.skip(128 * 2);
+#endif
+	}
+
+	int scoresLoaded = 0;
+	// uint16 number of Scores
+	const uint16 scoresInFile = musicData.readUint16BE();
+
+	if (musicData.err() || musicData.eos())
+		goto ioError;
+
+	if (loadScores) {
+		const uint16 tempScores = MIN(scoresInFile, _playerCtx.maxScoreNum);
+		Score *curScore = new Score[tempScores];
+		if (!curScore)
+			goto allocError;
+		_scores = curScore;
+
+		for (scoresLoaded = 0; scoresLoaded < tempScores; ++scoresLoaded, ++curScore) {
+			const uint32 numEvents = musicData.readUint32BE();
+			Event *curEvent = new Event[numEvents];
+			if (!curEvent)
+				goto allocError;
+			curScore->events = curEvent;
+			for (int j = numEvents; j > 0; --j, ++curEvent) {
+				curEvent->command = musicData.readByte();
+				curEvent->parameter = musicData.readByte();
+				curEvent->startTime = musicData.readUint16BE();
+				curEvent->stopTime = musicData.readUint16BE();
+			}
+			curScore->numEvents = numEvents;
+		}
+		_numScores = scoresLoaded;
+	}
+
+	if (loadSamples) {
+		// skip over remaining scores in file
+		for (int i = scoresInFile - scoresLoaded; i > 0; --i)
+			musicData.skip(musicData.readUint32BE() * 6);
+
+		// uint16 number of Samples
+		const uint16 wavesInFile = musicData.readUint16BE();
+		for (int i = wavesInFile; i > 0; --i) {
+			// load disksample structure
+			const uint16 number = musicData.readUint16BE();
+			assert(number < ARRAYSIZE(_patch));
+
+			Patch &curPatch = _patch[number];
+			if (curPatch.attackPtr || curPatch.samplePtr) {
+				delete curPatch.attackPtr;
+				curPatch.attackPtr = 0;
+				delete curPatch.samplePtr;
+				curPatch.samplePtr = 0;
+			}
+			curPatch.tune = musicData.readSint16BE();
+			curPatch.volume = musicData.readUint16BE();
+			curPatch.sampleOctaves = musicData.readUint16BE();
+			curPatch.sampleAttackLen = musicData.readUint32BE();
+			const uint32 sustainLen = musicData.readUint32BE();
+			curPatch.sampleTotalLen = curPatch.sampleAttackLen + sustainLen;
+			// each octave the number of samples doubles.
+			const uint32 totalSamples = curPatch.sampleTotalLen * ((1 << curPatch.sampleOctaves) - 1);
+			curPatch.attackLen = musicData.readUint16BE();
+			curPatch.releaseLen = musicData.readUint16BE();
+			const uint32 totalEnvs = curPatch.attackLen + curPatch.releaseLen;
+
+			// Allocate space for both attack and release Segment.
+			Envelope *envPtr = new Envelope[totalEnvs];
+			if (!envPtr)
+				goto allocError;
+			// Attack Segment
+			curPatch.attackPtr = envPtr;
+			// Release Segment
+			// curPatch.releasePtr = envPtr + curPatch.attackLen;
+
+			// Read Attack and Release Segments
+			for (int j = totalEnvs; j > 0; --j, ++envPtr) {
+				envPtr->duration = musicData.readUint16BE();
+				envPtr->volume = musicData.readUint16BE();
+			}
+
+			// read Samples
+			int8 *allocSamples = new int8[totalSamples];
+			if (!allocSamples)
+				goto allocError;
+			curPatch.samplePtr = allocSamples;
+			musicData.read(allocSamples, totalSamples);
+		}
+	}
+	if (!musicData.err() && !musicData.eos())
+		return true;
+ioError:
+	errorMsg = "Maxtrax: Encountered IO-Error";
+allocError:
+	if (!errorMsg)
+		errorMsg = "Maxtrax: Could not allocate Memory";
+
+	warning("%s", errorMsg);
+	if (loadSamples)
+		freePatches();
+	if (loadScores)
+		freeScores();
+	return false;
+}
+
+#if !defined(NDEBUG) && 0
+void MaxTrax::outPutEvent(const Event &ev, int num) {
+	struct {
+		byte cmd;
+		const char *name;
+		const char *param;
+	} COMMANDS[] = {
+		{0x80, "TEMPO   ", "TEMPO, N/A      "},
+		{0xa0, "SPECIAL ", "CHAN, SPEC # | VAL"},
+		{0xb0, "CONTROL ", "CHAN, CTRL # | VAL"},
+		{0xc0, "PROGRAM ", "CHANNEL, PROG # "},
+		{0xe0, "BEND    ", "CHANNEL, BEND VALUE"},
+		{0xf0, "SYSEX   ", "TYPE, SIZE      "},
+		{0xf8, "REALTIME", "REALTIME, N/A   "},
+		{0xff, "END     ", "N/A, N/A        "},
+		{0xff, "NOTE    ", "VOL | CHAN, STOP"},
+	};
+
+	int i = 0;
+	for (; i < ARRAYSIZE(COMMANDS) - 1 && ev.command != COMMANDS[i].cmd; ++i)
+		;
+
+	if (num == -1)
+		debug("Event    : %02X %s %s %02X %04X %04X", ev.command, COMMANDS[i].name, COMMANDS[i].param, ev.parameter, ev.startTime, ev.stopTime);
+	else
+		debug("Event %3d: %02X %s %s %02X %04X %04X", num, ev.command, COMMANDS[i].name, COMMANDS[i].param, ev.parameter, ev.startTime, ev.stopTime);
+}
+
+void MaxTrax::outPutScore(const Score &sc, int num) {
+	if (num == -1)
+		debug("score   : %i Events", sc.numEvents);
+	else
+		debug("score %2d: %i Events", num, sc.numEvents);
+	for (uint i = 0; i < sc.numEvents; ++i)
+		outPutEvent(sc.events[i], i);
+	debug("");
+}
+#else
+void MaxTrax::outPutEvent(const Event &ev, int num) {}
+void MaxTrax::outPutScore(const Score &sc, int num) {}
+#endif	// #ifndef NDEBUG
+
+}	// End of namespace Audio
+
+#endif // #if defined(SOUND_MODS_MAXTRAX_H)
diff --git a/audio/mods/maxtrax.h b/audio/mods/maxtrax.h
new file mode 100644
index 0000000000..2f890afe2d
--- /dev/null
+++ b/audio/mods/maxtrax.h
@@ -0,0 +1,225 @@
+/* ScummVM - Graphic Adventure Engine
+ *
+ * ScummVM is the legal property of its developers, whose names
+ * are too numerous to list here. Please refer to the COPYRIGHT
+ * file distributed with this source distribution.
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
+ *
+ * $URL$
+ * $Id$
+ *
+ */
+
+// see if all engines using this class are DISABLED
+#if !defined(ENABLE_KYRA)
+
+// normal Header Guard
+#elif !defined SOUND_MODS_MAXTRAX_H
+#define SOUND_MODS_MAXTRAX_H
+
+// #define MAXTRAX_HAS_MODULATION
+// #define MAXTRAX_HAS_MICROTONAL
+
+#include "audio/mods/paula.h"
+
+namespace Audio {
+
+class MaxTrax : public Paula {
+public:
+	MaxTrax(int rate, bool stereo, uint16 vBlankFreq = 50, uint16 maxScores = 128);
+	virtual ~MaxTrax();
+
+	bool load(Common::SeekableReadStream &musicData, bool loadScores = true, bool loadSamples = true);
+	bool playSong(int songIndex, bool loop = false);
+	void advanceSong(int advance = 1);
+	int playNote(byte note, byte patch, uint16 duration, uint16 volume, bool rightSide);
+	void setVolume(const byte volume) { Common::StackLock lock(_mutex); _playerCtx.volume = volume; }
+	void setTempo(const uint16 tempo);
+	void stopMusic();
+	/**
+	 * Set a callback function for sync-events.
+	 * @param callback Callback function, will be called synchronously, so DONT modify the player
+	 *		directly in response
+	 */
+	void setSignalCallback(void (*callback) (int));
+
+protected:
+	void interrupt();
+
+private:
+	enum { kNumPatches = 64, kNumVoices = 4, kNumChannels = 16, kNumExtraChannels = 1 };
+	enum { kPriorityScore, kPriorityNote, kPrioritySound };
+
+#ifdef MAXTRAX_HAS_MICROTONAL
+	int16	_microtonal[128];
+#endif
+
+	struct Event {
+		uint16	startTime;
+		uint16	stopTime;
+		byte	command;
+		byte	parameter;
+	};
+
+	const struct Score {
+		const Event	*events;
+		uint32	numEvents;
+	} *_scores;
+
+	int _numScores;
+
+	struct {
+		uint32	sineValue;
+		uint16	vBlankFreq;
+		int32	ticks;
+		int32	tickUnit;
+		uint16	frameUnit;
+
+		uint16	maxScoreNum;
+		uint16	tempo;
+		uint16	tempoInitial;
+		uint16	tempoStart;
+		int16	tempoDelta;
+		int32	tempoTime;
+		int32	tempoTicks;
+
+		byte	volume;
+
+		bool	filterOn;
+		bool	handleVolume;
+		bool	musicLoop;
+
+		int		scoreIndex;
+		const Event	*nextEvent;
+		int32	nextEventTime;
+
+		void (*syncCallBack) (int);
+		const Event	*repeatPoint[4];
+		byte	repeatCount[4];
+	} _playerCtx;
+
+	struct Envelope {
+		uint16	duration;
+		uint16	volume;
+	};
+
+	struct Patch {
+		const Envelope *attackPtr;
+		//Envelope *releasePtr;
+		uint16	attackLen;
+		uint16	releaseLen;
+
+		int16	tune;
+		uint16	volume;
+
+		// this was the SampleData struct in the assembler source
+		const int8	*samplePtr;
+		uint32	sampleTotalLen;
+		uint32	sampleAttackLen;
+		uint16	sampleOctaves;
+	} _patch[kNumPatches];
+
+	struct ChannelContext {
+		const Patch	*patch;
+		uint16	regParamNumber;
+
+		uint16	modulation;
+		uint16	modulationTime;
+
+		int16	microtonal;
+
+		uint16	portamentoTime;
+
+		int16	pitchBend;
+		int16	pitchReal;
+		int8	pitchBendRange;
+
+		uint8	volume;
+//		uint8	voicesActive;
+
+		enum {
+			kFlagRightChannel = 1 << 0,
+			kFlagPortamento = 1 << 1,
+			kFlagDamper = 1 << 2,
+			kFlagMono = 1 << 3,
+			kFlagMicrotonal = 1 << 4,
+			kFlagModVolume = 1 << 5
+		};
+		byte	flags;
+		bool	isAltered;
+
+		uint8	lastNote;
+//		uint8	program;
+
+	} _channelCtx[kNumChannels + kNumExtraChannels];
+
+	struct VoiceContext {
+		ChannelContext *channel;
+		const Patch	*patch;
+		const Envelope *envelope;
+//		uint32	uinqueId;
+		int32	preCalcNote;
+		uint32	ticksLeft;
+		int32	portaTicks;
+		int32	incrVolume;
+//		int32	periodOffset;
+		uint16	envelopeLeft;
+		uint16	noteVolume;
+		uint16	baseVolume;
+		uint16	lastPeriod;
+		byte	baseNote;
+		byte	endNote;
+		byte	octave;
+//		byte	number;
+//		byte	link;
+		enum {
+			kStatusFree,
+			kStatusHalt,
+			kStatusDecay,
+			kStatusRelease,
+			kStatusSustain,
+			kStatusAttack,
+			kStatusStart
+		};
+		uint8	isBlocked;
+		uint8	priority;
+		byte	status;
+		byte	lastVolume;
+		byte	tieBreak;
+		bool	hasDamper;
+		bool	hasPortamento;
+		byte	dmaOff;
+
+		int32	stopEventTime;
+	} _voiceCtx[kNumVoices];
+
+	void controlCh(ChannelContext &channel, byte command, byte data);
+	void freePatches();
+	void freeScores();
+	void resetChannel(ChannelContext &chan, bool rightChannel);
+	void resetPlayer();
+
+	int8 pickvoice(uint pick, int16 pri);
+	uint16 calcNote(const VoiceContext &voice);
+	int8 noteOn(ChannelContext &channel, byte note, uint16 volume, uint16 pri);
+	void killVoice(byte num);
+
+	static void outPutEvent(const Event &ev, int num = -1);
+	static void outPutScore(const Score &sc, int num = -1);
+};
+}	// End of namespace Audio
+
+#endif // !defined SOUND_MODS_MAXTRAX_H
diff --git a/audio/mods/module.cpp b/audio/mods/module.cpp
new file mode 100644
index 0000000000..0da6923b5d
--- /dev/null
+++ b/audio/mods/module.cpp
@@ -0,0 +1,252 @@
+/* ScummVM - Graphic Adventure Engine
+ *
+ * ScummVM is the legal property of its developers, whose names
+ * are too numerous to list here. Please refer to the COPYRIGHT
+ * file distributed with this source distribution.
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
+ *
+ * $URL$
+ * $Id$
+ *
+ */
+
+#include "audio/mods/module.h"
+
+#include "common/util.h"
+#include "common/endian.h"
+
+namespace Modules {
+
+const int16 Module::periods[16][60] = {
+	{1712, 1616, 1524, 1440, 1356, 1280, 1208, 1140, 1076, 1016, 960 , 906,
+	 856 , 808 , 762 , 720 , 678 , 640 , 604 , 570 , 538 , 508 , 480 , 453,
+	 428 , 404 , 381 , 360 , 339 , 320 , 302 , 285 , 269 , 254 , 240 , 226,
+	 214 , 202 , 190 , 180 , 170 , 160 , 151 , 143 , 135 , 127 , 120 , 113,
+	 107 , 101 , 95  , 90  , 85  , 80  , 75  , 71  , 67  , 63  , 60  , 56 },
+	{1700, 1604, 1514, 1430, 1348, 1274, 1202, 1134, 1070, 1010, 954 , 900,
+	 850 , 802 , 757 , 715 , 674 , 637 , 601 , 567 , 535 , 505 , 477 , 450,
+	 425 , 401 , 379 , 357 , 337 , 318 , 300 , 284 , 268 , 253 , 239 , 225,
+	 213 , 201 , 189 , 179 , 169 , 159 , 150 , 142 , 134 , 126 , 119 , 113,
+	 106 , 100 , 94  , 89  , 84  , 79  , 75  , 71  , 67  , 63  , 59  , 56 },
+	{1688, 1592, 1504, 1418, 1340, 1264, 1194, 1126, 1064, 1004, 948 , 894,
+	 844 , 796 , 752 , 709 , 670 , 632 , 597 , 563 , 532 , 502 , 474 , 447,
+	 422 , 398 , 376 , 355 , 335 , 316 , 298 , 282 , 266 , 251 , 237 , 224,
+	 211 , 199 , 188 , 177 , 167 , 158 , 149 , 141 , 133 , 125 , 118 , 112,
+	 105 , 99  , 94  , 88  , 83  , 79  , 74  , 70  , 66  , 62  , 59  , 56 },
+	{1676, 1582, 1492, 1408, 1330, 1256, 1184, 1118, 1056, 996 , 940 , 888,
+	 838 , 791 , 746 , 704 , 665 , 628 , 592 , 559 , 528 , 498 , 470 , 444,
+	 419 , 395 , 373 , 352 , 332 , 314 , 296 , 280 , 264 , 249 , 235 , 222,
+	 209 , 198 , 187 , 176 , 166 , 157 , 148 , 140 , 132 , 125 , 118 , 111,
+	 104 , 99  , 93  , 88  , 83  , 78  , 74  , 70  , 66  , 62  , 59  , 55 },
+	{1664, 1570, 1482, 1398, 1320, 1246, 1176, 1110, 1048, 990 , 934 , 882,
+	 832 , 785 , 741 , 699 , 660 , 623 , 588 , 555 , 524 , 495 , 467 , 441,
+	 416 , 392 , 370 , 350 , 330 , 312 , 294 , 278 , 262 , 247 , 233 , 220,
+	 208 , 196 , 185 , 175 , 165 , 156 , 147 , 139 , 131 , 124 , 117 , 110,
+	 104 , 98  , 92  , 87  , 82  , 78  , 73  , 69  , 65  , 62  , 58  , 55 },
+	{1652, 1558, 1472, 1388, 1310, 1238, 1168, 1102, 1040, 982 , 926 , 874,
+	 826 , 779 , 736 , 694 , 655 , 619 , 584 , 551 , 520 , 491 , 463 , 437,
+	 413 , 390 , 368 , 347 , 328 , 309 , 292 , 276 , 260 , 245 , 232 , 219,
+	 206 , 195 , 184 , 174 , 164 , 155 , 146 , 138 , 130 , 123 , 116 , 109,
+	 103 , 97  , 92  , 87  , 82  , 77  , 73  , 69  , 65  , 61  , 58  , 54 },
+	{1640, 1548, 1460, 1378, 1302, 1228, 1160, 1094, 1032, 974 , 920 , 868,
+	 820 , 774 , 730 , 689 , 651 , 614 , 580 , 547 , 516 , 487 , 460 , 434,
+	 410 , 387 , 365 , 345 , 325 , 307 , 290 , 274 , 258 , 244 , 230 , 217,
+	 205 , 193 , 183 , 172 , 163 , 154 , 145 , 137 , 129 , 122 , 115 , 109,
+	 102 , 96  , 91  , 86  , 81  , 77  , 72  , 68  , 64  , 61  , 57  , 54 },
+	{1628, 1536, 1450, 1368, 1292, 1220, 1150, 1086, 1026, 968 , 914 , 862,
+	 814 , 768 , 725 , 684 , 646 , 610 , 575 , 543 , 513 , 484 , 457 , 431,
+	 407 , 384 , 363 , 342 , 323 , 305 , 288 , 272 , 256 , 242 , 228 , 216,
+	 204 , 192 , 181 , 171 , 161 , 152 , 144 , 136 , 128 , 121 , 114 , 108,
+	 102 , 96  , 90  , 85  , 80  , 76  , 72  , 68  , 64  , 60  , 57  , 54 },
+	{1814, 1712, 1616, 1524, 1440, 1356, 1280, 1208, 1140, 1076, 1016, 960,
+	 907 , 856 , 808 , 762 , 720 , 678 , 640 , 604 , 570 , 538 , 508 , 480,
+	 453 , 428 , 404 , 381 , 360 , 339 , 320 , 302 , 285 , 269 , 254 , 240,
+	 226 , 214 , 202 , 190 , 180 , 170 , 160 , 151 , 143 , 135 , 127 , 120,
+	 113 , 107 , 101 , 95  , 90  , 85  , 80  , 75  , 71  , 67  , 63  , 60 },
+	{1800, 1700, 1604, 1514, 1430, 1350, 1272, 1202, 1134, 1070, 1010, 954,
+	 900 , 850 , 802 , 757 , 715 , 675 , 636 , 601 , 567 , 535 , 505 , 477,
+	 450 , 425 , 401 , 379 , 357 , 337 , 318 , 300 , 284 , 268 , 253 , 238,
+	 225 , 212 , 200 , 189 , 179 , 169 , 159 , 150 , 142 , 134 , 126 , 119,
+	 112 , 106 , 100 , 94  , 89  , 84  , 79  , 75  , 71  , 67  , 63  , 59 },
+	{1788, 1688, 1592, 1504, 1418, 1340, 1264, 1194, 1126, 1064, 1004, 948,
+	 894 , 844 , 796 , 752 , 709 , 670 , 632 , 597 , 563 , 532 , 502 , 474,
+	 447 , 422 , 398 , 376 , 355 , 335 , 316 , 298 , 282 , 266 , 251 , 237,
+	 223 , 211 , 199 , 188 , 177 , 167 , 158 , 149 , 141 , 133 , 125 , 118,
+	 111 , 105 , 99  , 94  , 88  , 83  , 79  , 74  , 70  , 66  , 62  , 59 },
+	{1774, 1676, 1582, 1492, 1408, 1330, 1256, 1184, 1118, 1056, 996 , 940,
+	 887 , 838 , 791 , 746 , 704 , 665 , 628 , 592 , 559 , 528 , 498 , 470,
+	 444 , 419 , 395 , 373 , 352 , 332 , 314 , 296 , 280 , 264 , 249 , 235,
+	 222 , 209 , 198 , 187 , 176 , 166 , 157 , 148 , 140 , 132 , 125 , 118,
+	 111 , 104 , 99  , 93  , 88  , 83  , 78  , 74  , 70  , 66  , 62  , 59 },
+	{1762, 1664, 1570, 1482, 1398, 1320, 1246, 1176, 1110, 1048, 988 , 934,
+	 881 , 832 , 785 , 741 , 699 , 660 , 623 , 588 , 555 , 524 , 494 , 467,
+	 441 , 416 , 392 , 370 , 350 , 330 , 312 , 294 , 278 , 262 , 247 , 233,
+	 220 , 208 , 196 , 185 , 175 , 165 , 156 , 147 , 139 , 131 , 123 , 117,
+	 110 , 104 , 98  , 92  , 87  , 82  , 78  , 73  , 69  , 65  , 61  , 58 },
+	{1750, 1652, 1558, 1472, 1388, 1310, 1238, 1168, 1102, 1040, 982 , 926,
+	 875 , 826 , 779 , 736 , 694 , 655 , 619 , 584 , 551 , 520 , 491 , 463,
+	 437 , 413 , 390 , 368 , 347 , 328 , 309 , 292 , 276 , 260 , 245 , 232,
+	 219 , 206 , 195 , 184 , 174 , 164 , 155 , 146 , 138 , 130 , 123 , 116,
+	 109 , 103 , 97  , 92  , 87  , 82  , 77  , 73  , 69  , 65  , 61  , 58 },
+	{1736, 1640, 1548, 1460, 1378, 1302, 1228, 1160, 1094, 1032, 974 , 920,
+	 868 , 820 , 774 , 730 , 689 , 651 , 614 , 580 , 547 , 516 , 487 , 460,
+	 434 , 410 , 387 , 365 , 345 , 325 , 307 , 290 , 274 , 258 , 244 , 230,
+	 217 , 205 , 193 , 183 , 172 , 163 , 154 , 145 , 137 , 129 , 122 , 115,
+	 108 , 102 , 96  , 91  , 86  , 81  , 77  , 72  , 68  , 64  , 61  , 57 },
+	{1724, 1628, 1536, 1450, 1368, 1292, 1220, 1150, 1086, 1026, 968 , 914,
+	 862 , 814 , 768 , 725 , 684 , 646 , 610 , 575 , 543 , 513 , 484 , 457,
+	 431 , 407 , 384 , 363 , 342 , 323 , 305 , 288 , 272 , 256 , 242 , 228,
+	 216 , 203 , 192 , 181 , 171 , 161 , 152 , 144 , 136 , 128 , 121 , 114,
+	 108 , 101 , 96  , 90  , 85  , 80  , 76  , 72  , 68  , 64  , 60  , 57 }};
+
+const uint32 Module::signatures[] = {
+	MKID_BE('M.K.'), MKID_BE('M!K!'), MKID_BE('FLT4')
+};
+
+bool Module::load(Common::SeekableReadStream &st, int offs) {
+	if (offs) {
+		// Load the module with the common sample data
+		load(st, 0);
+	}
+
+	st.seek(offs);
+	st.read(songname, 20);
+	songname[20] = '\0';
+
+	for (int i = 0; i < NUM_SAMPLES; ++i) {
+		st.read(sample[i].name, 22);
+		sample[i].name[22] = '\0';
+		sample[i].len = 2 * st.readUint16BE();
+
+		sample[i].finetune = st.readByte();
+		assert(sample[i].finetune < 0x10);
+
+		sample[i].vol = st.readByte();
+		sample[i].repeat = 2 * st.readUint16BE();
+		sample[i].replen = 2 * st.readUint16BE();
+	}
+
+	songlen = st.readByte();
+	undef = st.readByte();
+
+	st.read(songpos, 128);
+
+	sig = st.readUint32BE();
+
+	bool foundSig = false;
+	for (int i = 0; i < ARRAYSIZE(signatures); i++) {
+		if (sig == signatures[i]) {
+			foundSig = true;
+			break;
+		}
+	}
+
+	if (!foundSig) {
+		warning("No known signature found in protracker module");
+		return false;
+	}
+
+	int maxpattern = 0;
+	for (int i = 0; i < 128; ++i)
+		if (maxpattern < songpos[i])
+			maxpattern = songpos[i];
+
+	pattern = new pattern_t[maxpattern + 1];
+
+	for (int i = 0; i <= maxpattern; ++i) {
+		for (int j = 0; j < 64; ++j) {
+			for (int k = 0; k < 4; ++k) {
+				uint32 note = st.readUint32BE();
+				pattern[i][j][k].sample = (note & 0xf0000000) >> 24 | (note & 0x0000f000) >> 12;
+				pattern[i][j][k].period = (note >> 16) & 0xfff;
+				pattern[i][j][k].effect = note & 0xfff;
+				pattern[i][j][k].note = periodToNote((note >> 16) & 0xfff);
+			}
+		}
+	}
+
+	for (int i = 0; i < NUM_SAMPLES; ++i) {
+		if (offs) {
+			// Restore information for modules that use common sample data
+			for (int j = 0; j < NUM_SAMPLES; ++j) {
+				if (!scumm_stricmp((const char *)commonSamples[j].name, (const char *)sample[i].name)) {
+					sample[i].len = commonSamples[j].len;
+					st.seek(commonSamples[j].offs);
+					break;
+				}
+			}
+		} else {
+			// Store information for modules that use common sample data
+			memcpy(commonSamples[i].name, sample[i].name, 22);
+			commonSamples[i].len = sample[i].len;
+			commonSamples[i].offs = st.pos();
+
+		}
+
+		if (!sample[i].len) {
+			sample[i].data = 0;
+		} else {
+			sample[i].data = new int8[sample[i].len];
+			st.read((byte *)sample[i].data, sample[i].len);
+		}
+	}
+
+	return true;
+}
+
+Module::Module() {
+	pattern = 0;
+	for (int i = 0; i < NUM_SAMPLES; ++i) {
+		sample[i].data = 0;
+	}
+}
+
+Module::~Module() {
+	delete[] pattern;
+	for (int i = 0; i < NUM_SAMPLES; ++i) {
+		delete[] sample[i].data;
+	}
+}
+
+byte Module::periodToNote(int16 period, byte finetune) {
+	int16 diff1;
+	int16 diff2;
+
+	diff1 = ABS(periods[finetune][0] - period);
+	if (diff1 == 0)
+		return 0;
+
+	for (int i = 1; i < 60; i++) {
+		diff2 = ABS(periods[finetune][i] - period);
+		if (diff2 == 0)
+			return i;
+		else if (diff2 > diff1)
+			return i-1;
+		diff1 = diff2;
+	}
+	return 59;
+}
+
+int16 Module::noteToPeriod(byte note, byte finetune) {
+	if (finetune > 15)
+		finetune = 15;
+	if (note > 59)
+		note = 59;
+
+	return periods[finetune][note];
+}
+
+} // End of namespace Modules
diff --git a/audio/mods/module.h b/audio/mods/module.h
new file mode 100644
index 0000000000..550b63617e
--- /dev/null
+++ b/audio/mods/module.h
@@ -0,0 +1,90 @@
+/* ScummVM - Graphic Adventure Engine
+ *
+ * ScummVM is the legal property of its developers, whose names
+ * are too numerous to list here. Please refer to the COPYRIGHT
+ * file distributed with this source distribution.
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
+ *
+ * $URL$
+ * $Id$
+ *
+ */
+
+#ifndef SOUND_MODS_MODULE_H
+#define SOUND_MODS_MODULE_H
+
+#include "common/stream.h"
+
+namespace Modules {
+
+#include "common/pack-start.h"	// START STRUCT PACKING
+
+struct note_t {
+	byte sample;
+	byte note;
+	uint16 period;
+	uint16 effect;
+} PACKED_STRUCT;
+
+#include "common/pack-end.h"	// END STRUCT PACKING
+
+typedef note_t pattern_t[64][4];
+
+struct sample_t {
+	byte name[23];
+	uint16 len;
+	byte finetune;
+	byte vol;
+	uint16 repeat;
+	uint16 replen;
+	int8 *data;
+};
+
+struct sample_offs {
+	byte name[23];
+	uint16 len;
+	uint32 offs;
+};
+
+class Module {
+public:
+	byte songname[21];
+
+	static const int NUM_SAMPLES = 31;
+	sample_t sample[NUM_SAMPLES];
+	sample_offs commonSamples[NUM_SAMPLES];
+
+	byte songlen;
+	byte undef;
+	byte songpos[128];
+	uint32 sig;
+	pattern_t *pattern;
+
+	Module();
+	~Module();
+
+	bool load(Common::SeekableReadStream &stream, int offs);
+	static byte periodToNote(int16 period, byte finetune = 0);
+	static int16 noteToPeriod(byte note, byte finetune = 0);
+
+private:
+	static const int16 periods[16][60];
+	static const uint32 signatures[];
+};
+
+} // End of namespace Modules
+
+#endif
diff --git a/audio/mods/paula.cpp b/audio/mods/paula.cpp
new file mode 100644
index 0000000000..ef841ac9bf
--- /dev/null
+++ b/audio/mods/paula.cpp
@@ -0,0 +1,212 @@
+/* ScummVM - Graphic Adventure Engine
+ *
+ * ScummVM is the legal property of its developers, whose names
+ * are too numerous to list here. Please refer to the COPYRIGHT
+ * file distributed with this source distribution.
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
+ *
+ * $URL$
+ * $Id$
+ *
+ */
+
+#include "audio/mods/paula.h"
+#include "audio/null.h"
+
+namespace Audio {
+
+Paula::Paula(bool stereo, int rate, uint interruptFreq) :
+		_stereo(stereo), _rate(rate), _periodScale((double)kPalPaulaClock / rate), _intFreq(interruptFreq) {
+
+	clearVoices();
+	_voice[0].panning = 191;
+	_voice[1].panning = 63;
+	_voice[2].panning = 63;
+	_voice[3].panning = 191;
+
+	if (_intFreq == 0)
+		_intFreq = _rate;
+
+	_curInt = 0;
+	_timerBase = 1;
+	_playing = false;
+	_end = true;
+}
+
+Paula::~Paula() {
+}
+
+void Paula::clearVoice(byte voice) {
+	assert(voice < NUM_VOICES);
+
+	_voice[voice].data = 0;
+	_voice[voice].dataRepeat = 0;
+	_voice[voice].length = 0;
+	_voice[voice].lengthRepeat = 0;
+	_voice[voice].period = 0;
+	_voice[voice].volume = 0;
+	_voice[voice].offset = Offset(0);
+	_voice[voice].dmaCount = 0;
+}
+
+int Paula::readBuffer(int16 *buffer, const int numSamples) {
+	Common::StackLock lock(_mutex);
+
+	memset(buffer, 0, numSamples * 2);
+	if (!_playing) {
+		return numSamples;
+	}
+
+	if (_stereo)
+		return readBufferIntern<true>(buffer, numSamples);
+	else
+		return readBufferIntern<false>(buffer, numSamples);
+}
+
+
+template<bool stereo>
+inline int mixBuffer(int16 *&buf, const int8 *data, Paula::Offset &offset, frac_t rate, int neededSamples, uint bufSize, byte volume, byte panning) {
+	int samples;
+	for (samples = 0; samples < neededSamples && offset.int_off < bufSize; ++samples) {
+		const int32 tmp = ((int32) data[offset.int_off]) * volume;
+		if (stereo) {
+			*buf++ += (tmp * (255 - panning)) >> 7;
+			*buf++ += (tmp * (panning)) >> 7;
+		} else
+			*buf++ += tmp;
+
+		// Step to next source sample
+		offset.rem_off += rate;
+		if (offset.rem_off >= (frac_t)FRAC_ONE) {
+			offset.int_off += fracToInt(offset.rem_off);
+			offset.rem_off &= FRAC_LO_MASK;
+		}
+	}
+
+	return samples;
+}
+
+template<bool stereo>
+int Paula::readBufferIntern(int16 *buffer, const int numSamples) {
+	int samples = _stereo ? numSamples / 2 : numSamples;
+	while (samples > 0) {
+
+		// Handle 'interrupts'. This gives subclasses the chance to adjust the channel data
+		// (e.g. insert new samples, do pitch bending, whatever).
+		if (_curInt == 0) {
+			_curInt = _intFreq;
+			interrupt();
+		}
+
+		// Compute how many samples to generate: at most the requested number of samples,
+		// of course, but we may stop earlier when an 'interrupt' is expected.
+		const uint nSamples = MIN((uint)samples, _curInt);
+
+		// Loop over the four channels of the emulated Paula chip
+		for (int voice = 0; voice < NUM_VOICES; voice++) {
+			// No data, or paused -> skip channel
+			if (!_voice[voice].data || (_voice[voice].period <= 0))
+				continue;
+
+			// The Paula chip apparently run at 7.0937892 MHz in the PAL
+			// version and at 7.1590905 MHz in the NTSC version. We divide this
+			// by the requested the requested output sampling rate _rate
+			// (typically 44.1 kHz or 22.05 kHz) obtaining the value _periodScale.
+			// This is then divided by the "period" of the channel we are
+			// processing, to obtain the correct output 'rate'.
+			frac_t rate = doubleToFrac(_periodScale / _voice[voice].period);
+			// Cap the volume
+			_voice[voice].volume = MIN((byte) 0x40, _voice[voice].volume);
+
+
+			Channel &ch = _voice[voice];
+			int16 *p = buffer;
+			int neededSamples = nSamples;
+			assert(ch.offset.int_off < ch.length);
+
+			// Mix the generated samples into the output buffer
+			neededSamples -= mixBuffer<stereo>(p, ch.data, ch.offset, rate, neededSamples, ch.length, ch.volume, ch.panning);
+
+			// Wrap around if necessary
+			if (ch.offset.int_off >= ch.length) {
+				// Important: Wrap around the offset *before* updating the voice length.
+				// Otherwise, if length != lengthRepeat we would wrap incorrectly.
+				// Note: If offset >= 2*len ever occurs, the following would be wrong;
+				// instead of subtracting, we then should compute the modulus using "%=".
+				// Since that requires a division and is slow, and shouldn't be necessary
+				// in practice anyway, we only use subtraction.
+				ch.offset.int_off -= ch.length;
+				ch.dmaCount++;
+
+				ch.data = ch.dataRepeat;
+				ch.length = ch.lengthRepeat;
+			}
+
+			// If we have not yet generated enough samples, and looping is active: loop!
+			if (neededSamples > 0 && ch.length > 2) {
+				// Repeat as long as necessary.
+				while (neededSamples > 0) {
+					// Mix the generated samples into the output buffer
+					neededSamples -= mixBuffer<stereo>(p, ch.data, ch.offset, rate, neededSamples, ch.length, ch.volume, ch.panning);
+
+					if (ch.offset.int_off >= ch.length) {
+						// Wrap around. See also the note above.
+						ch.offset.int_off -= ch.length;
+						ch.dmaCount++;
+					}
+				}
+			}
+
+		}
+		buffer += _stereo ? nSamples * 2 : nSamples;
+		_curInt -= nSamples;
+		samples -= nSamples;
+	}
+	return numSamples;
+}
+
+} // End of namespace Audio
+
+
+//	Plugin interface
+//	(This can only create a null driver since apple II gs support seeems not to be implemented
+//  and also is not part of the midi driver architecture. But we need the plugin for the options
+//  menu in the launcher and for MidiDriver::detectDevice() which is more or less used by all engines.)
+
+class AmigaMusicPlugin : public NullMusicPlugin {
+public:
+	const char *getName() const {
+		return _s("Amiga Audio Emulator");
+	}
+
+	const char *getId() const {
+		return "amiga";
+	}
+
+	MusicDevices getDevices() const;
+};
+
+MusicDevices AmigaMusicPlugin::getDevices() const {
+	MusicDevices devices;
+	devices.push_back(MusicDevice(this, "", MT_AMIGA));
+	return devices;
+}
+
+//#if PLUGIN_ENABLED_DYNAMIC(AMIGA)
+	//REGISTER_PLUGIN_DYNAMIC(AMIGA, PLUGIN_TYPE_MUSIC, AmigaMusicPlugin);
+//#else
+	REGISTER_PLUGIN_STATIC(AMIGA, PLUGIN_TYPE_MUSIC, AmigaMusicPlugin);
+//#endif
diff --git a/audio/mods/paula.h b/audio/mods/paula.h
new file mode 100644
index 0000000000..f6f159d5a6
--- /dev/null
+++ b/audio/mods/paula.h
@@ -0,0 +1,210 @@
+/* ScummVM - Graphic Adventure Engine
+ *
+ * ScummVM is the legal property of its developers, whose names
+ * are too numerous to list here. Please refer to the COPYRIGHT
+ * file distributed with this source distribution.
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
+ *
+ * $URL$
+ * $Id$
+ *
+ */
+
+#ifndef SOUND_MODS_PAULA_H
+#define SOUND_MODS_PAULA_H
+
+#include "audio/audiostream.h"
+#include "common/frac.h"
+#include "common/mutex.h"
+
+namespace Audio {
+
+/**
+ * Emulation of the "Paula" Amiga music chip
+ * The interrupt frequency specifies the number of mixed wavesamples between
+ * calls of the interrupt method
+ */
+class Paula : public AudioStream {
+public:
+	static const int NUM_VOICES = 4;
+	enum {
+		kPalSystemClock  = 7093790,
+		kNtscSystemClock = 7159090,
+		kPalCiaClock     = kPalSystemClock / 10,
+		kNtscCiaClock    = kNtscSystemClock / 10,
+		kPalPaulaClock   = kPalSystemClock / 2,
+		kNtscPauleClock  = kNtscSystemClock / 2
+	};
+
+	/* TODO: Document this */
+	struct Offset {
+		uint	int_off;	// integral part of the offset
+		frac_t	rem_off;	// fractional part of the offset, at least 0 and less than 1
+
+		explicit Offset(int off = 0) : int_off(off), rem_off(0) {}
+	};
+
+	Paula(bool stereo = false, int rate = 44100, uint interruptFreq = 0);
+	~Paula();
+
+	bool playing() const { return _playing; }
+	void setTimerBaseValue( uint32 ticksPerSecond ) { _timerBase = ticksPerSecond; }
+	uint32 getTimerBaseValue() { return _timerBase; }
+	void setSingleInterrupt(uint sampleDelay) { assert(sampleDelay < _intFreq); _curInt = sampleDelay; }
+	void setSingleInterruptUnscaled(uint timerDelay) {
+		setSingleInterrupt((uint)(((double)timerDelay * getRate()) / _timerBase));
+	}
+	void setInterruptFreq(uint sampleDelay) { _intFreq = sampleDelay; _curInt = 0; }
+	void setInterruptFreqUnscaled(uint timerDelay) {
+		setInterruptFreq((uint)(((double)timerDelay * getRate()) / _timerBase));
+	}
+	void clearVoice(byte voice);
+	void clearVoices() { for (int i = 0; i < NUM_VOICES; ++i) clearVoice(i); }
+	void startPlay() { _playing = true; }
+	void stopPlay() { _playing = false; }
+	void pausePlay(bool pause) { _playing = !pause; }
+
+// AudioStream API
+	int readBuffer(int16 *buffer, const int numSamples);
+	bool isStereo() const { return _stereo; }
+	bool endOfData() const { return _end; }
+	int getRate() const { return _rate; }
+
+protected:
+	struct Channel {
+		const int8 *data;
+		const int8 *dataRepeat;
+		uint32 length;
+		uint32 lengthRepeat;
+		int16 period;
+		byte volume;
+		Offset offset;
+		byte panning; // For stereo mixing: 0 = far left, 255 = far right
+		int dmaCount;
+	};
+
+	bool _end;
+	Common::Mutex _mutex;
+
+	virtual void interrupt() = 0;
+
+	void startPaula() {
+		_playing = true;
+		_end = false;
+	}
+
+	void stopPaula() {
+		_playing = false;
+		_end = true;
+	}
+
+	void setChannelPanning(byte channel, byte panning) {
+		assert(channel < NUM_VOICES);
+		_voice[channel].panning = panning;
+	}
+
+	void disableChannel(byte channel) {
+		assert(channel < NUM_VOICES);
+		_voice[channel].data = 0;
+	}
+
+	void enableChannel(byte channel) {
+		assert(channel < NUM_VOICES);
+		Channel &ch = _voice[channel];
+		ch.data = ch.dataRepeat;
+		ch.length = ch.lengthRepeat;
+		// actually first 2 bytes are dropped?
+		ch.offset = Offset(0);
+		// ch.period = ch.periodRepeat;
+	}
+
+	void setChannelPeriod(byte channel, int16 period) {
+		assert(channel < NUM_VOICES);
+		_voice[channel].period = period;
+	}
+
+	void setChannelVolume(byte channel, byte volume) {
+		assert(channel < NUM_VOICES);
+		_voice[channel].volume = volume;
+	}
+
+	void setChannelSampleStart(byte channel, const int8 *data) {
+		assert(channel < NUM_VOICES);
+		_voice[channel].dataRepeat = data;
+	}
+
+	void setChannelSampleLen(byte channel, uint32 length) {
+		assert(channel < NUM_VOICES);
+		assert(length < 32768/2);
+		_voice[channel].lengthRepeat = 2 * length;
+	}
+
+	void setChannelData(uint8 channel, const int8 *data, const int8 *dataRepeat, uint32 length, uint32 lengthRepeat, int32 offset = 0) {
+		assert(channel < NUM_VOICES);
+
+		Channel &ch = _voice[channel];
+
+		ch.dataRepeat = data;
+		ch.lengthRepeat = length;
+		enableChannel(channel);
+		ch.offset = Offset(offset);
+
+		ch.dataRepeat = dataRepeat;
+		ch.lengthRepeat = lengthRepeat;
+	}
+
+	void setChannelOffset(byte channel, Offset offset) {
+		assert(channel < NUM_VOICES);
+		_voice[channel].offset = offset;
+	}
+
+	Offset getChannelOffset(byte channel) {
+		assert(channel < NUM_VOICES);
+		return _voice[channel].offset;
+	}
+
+	int getChannelDmaCount(byte channel) {
+		assert(channel < NUM_VOICES);
+		return _voice[channel].dmaCount;
+	}
+
+	void setChannelDmaCount(byte channel, int dmaVal = 0) {
+		assert(channel < NUM_VOICES);
+		_voice[channel].dmaCount = dmaVal;
+	}
+
+	void setAudioFilter(bool enable) {
+		// TODO: implement
+	}
+
+private:
+	Channel _voice[NUM_VOICES];
+
+	const bool _stereo;
+	const int _rate;
+	const double _periodScale;
+	uint _intFreq;
+	uint _curInt;
+	uint32 _timerBase;
+	bool _playing;
+
+	template<bool stereo>
+	int readBufferIntern(int16 *buffer, const int numSamples);
+};
+
+} // End of namespace Audio
+
+#endif
diff --git a/audio/mods/protracker.cpp b/audio/mods/protracker.cpp
new file mode 100644
index 0000000000..6051338900
--- /dev/null
+++ b/audio/mods/protracker.cpp
@@ -0,0 +1,466 @@
+/* ScummVM - Graphic Adventure Engine
+ *
+ * ScummVM is the legal property of its developers, whose names
+ * are too numerous to list here. Please refer to the COPYRIGHT
+ * file distributed with this source distribution.
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
+ *
+ * $URL$
+ * $Id$
+ *
+ */
+
+#include "audio/mods/protracker.h"
+#include "audio/mods/paula.h"
+#include "audio/mods/module.h"
+
+#include "audio/audiostream.h"
+
+namespace Modules {
+
+class ProtrackerStream : public ::Audio::Paula {
+private:
+	Module _module;
+
+	int _tick;
+	int _row;
+	int _pos;
+
+	int _speed;
+	int _bpm;
+
+	// For effect 0xB - Jump To Pattern;
+	bool _hasJumpToPattern;
+	int _jumpToPattern;
+
+	// For effect 0xD - PatternBreak;
+	bool _hasPatternBreak;
+	int _skipRow;
+
+	// For effect 0xE6 - Pattern Loop
+	bool _hasPatternLoop;
+	int _patternLoopCount;
+	int _patternLoopRow;
+
+	// For effect 0xEE - Pattern Delay
+	byte _patternDelay;
+
+	static const int16 sinetable[];
+
+	struct {
+		byte sample;
+		uint16 period;
+		Offset offset;
+
+		byte vol;
+		byte finetune;
+
+		// For effect 0x0 - Arpeggio
+		bool arpeggio;
+		byte arpeggioNotes[3];
+
+		// For effect 0x3 - Porta to note
+		uint16 portaToNote;
+		byte portaToNoteSpeed;
+
+		// For effect 0x4 - Vibrato
+		int vibrato;
+		byte vibratoPos;
+		byte vibratoSpeed;
+		byte vibratoDepth;
+
+		// For effect 0xED - Delay sample
+		byte delaySample;
+		byte delaySampleTick;
+	} _track[4];
+
+public:
+	ProtrackerStream(Common::SeekableReadStream *stream, int offs, int rate, bool stereo);
+
+private:
+	void interrupt();
+
+	void doPorta(int track) {
+		if (_track[track].portaToNote && _track[track].portaToNoteSpeed) {
+			if (_track[track].period < _track[track].portaToNote) {
+				_track[track].period += _track[track].portaToNoteSpeed;
+				if (_track[track].period > _track[track].portaToNote)
+					_track[track].period = _track[track].portaToNote;
+			} else if (_track[track].period > _track[track].portaToNote) {
+				_track[track].period -= _track[track].portaToNoteSpeed;
+				if (_track[track].period < _track[track].portaToNote)
+					_track[track].period = _track[track].portaToNote;
+			}
+		}
+	}
+	void doVibrato(int track) {
+		_track[track].vibrato =
+				(_track[track].vibratoDepth * sinetable[_track[track].vibratoPos]) / 128;
+		_track[track].vibratoPos += _track[track].vibratoSpeed;
+		_track[track].vibratoPos %= 64;
+	}
+	void doVolSlide(int track, byte ex, byte ey) {
+		int vol = _track[track].vol;
+		if (ex == 0)
+			vol -= ey;
+		else if (ey == 0)
+			vol += ex;
+
+		if (vol < 0)
+			vol = 0;
+		else if (vol > 64)
+			vol = 64;
+
+		_track[track].vol = vol;
+	}
+
+	void updateRow();
+	void updateEffects();
+
+};
+
+const int16 ProtrackerStream::sinetable[64] = {
+		 0,   24,   49,   74,   97,  120,  141,  161,
+	 180,  197,  212,  224,  235,  244,  250,  253,
+	 255,  253,  250,  244,  235,  224,  212,  197,
+	 180,  161,  141,  120,   97,   74,   49,   24,
+		 0,  -24,  -49,  -74,  -97, -120, -141, -161,
+	-180, -197, -212, -224, -235, -244, -250, -253,
+	-255, -253, -250, -244, -235, -224, -212, -197,
+	-180, -161, -141, -120,  -97,  -74,  -49,  -24
+};
+
+ProtrackerStream::ProtrackerStream(Common::SeekableReadStream *stream, int offs, int rate, bool stereo) :
+		Paula(stereo, rate, rate/50) {
+	bool result = _module.load(*stream, offs);
+	assert(result);
+
+	_tick = _row = _pos = 0;
+
+	_speed = 6;
+	_bpm = 125;
+
+	_hasJumpToPattern = false;
+	_jumpToPattern = 0;
+
+	_hasPatternBreak = false;
+	_skipRow = 0;
+
+	_hasPatternLoop = false;
+	_patternLoopCount = 0;
+	_patternLoopRow = 0;
+
+	_patternDelay = 0;
+
+	memset(_track, 0, sizeof(_track));
+
+	startPaula();
+}
+
+void ProtrackerStream::updateRow() {
+	for (int track = 0; track < 4; track++) {
+		_track[track].arpeggio = false;
+		_track[track].vibrato = 0;
+		_track[track].delaySampleTick = 0;
+		const note_t note =
+		    _module.pattern[_module.songpos[_pos]][_row][track];
+
+		const int effect = note.effect >> 8;
+
+		if (note.sample) {
+			if (_track[track].sample != note.sample) {
+				_track[track].vibratoPos = 0;
+			}
+			_track[track].sample = note.sample;
+			_track[track].finetune = _module.sample[note.sample - 1].finetune;
+			_track[track].vol = _module.sample[note.sample - 1].vol;
+		}
+
+		if (note.period) {
+			if (effect != 3 && effect != 5) {
+				if (_track[track].finetune)
+					_track[track].period = _module.noteToPeriod(note.note, _track[track].finetune);
+				else
+					_track[track].period = note.period;
+				_track[track].offset = Offset(0);
+			}
+		}
+
+		const byte exy = note.effect & 0xff;
+		const byte ex = (note.effect >> 4) & 0xf;
+		const byte ey = note.effect & 0xf;
+
+		int vol;
+		switch (effect) {
+		case 0x0:
+			if (exy) {
+				_track[track].arpeggio = true;
+				if (note.period) {
+					_track[track].arpeggioNotes[0] = note.note;
+					_track[track].arpeggioNotes[1] = note.note + ex;
+					_track[track].arpeggioNotes[2] = note.note + ey;
+				}
+			}
+			break;
+		case 0x1:
+			break;
+		case 0x2:
+			break;
+		case 0x3:
+			if (note.period)
+				_track[track].portaToNote = note.period;
+			if (exy)
+				_track[track].portaToNoteSpeed = exy;
+			break;
+		case 0x4:
+			if (exy) {
+				_track[track].vibratoSpeed = ex;
+				_track[track].vibratoDepth = ey;
+			}
+			break;
+		case 0x5:
+			doPorta(track);
+			doVolSlide(track, ex, ey);
+			break;
+		case 0x6:
+			doVibrato(track);
+			doVolSlide(track, ex, ey);
+			break;
+		case 0x9: // Set sample offset
+			if (exy) {
+				_track[track].offset = Offset(exy * 256);
+				setChannelOffset(track, _track[track].offset);
+			}
+			break;
+		case 0xA:
+			break;
+		case 0xB:
+			_hasJumpToPattern = true;
+			_jumpToPattern = exy;
+			break;
+		case 0xC:
+			_track[track].vol = exy;
+			break;
+		case 0xD:
+			_hasPatternBreak = true;
+			_skipRow = ex * 10 + ey;
+			break;
+		case 0xE:
+			switch (ex) {
+			case 0x0: // Switch filters off
+				break;
+			case 0x1: // Fine slide up
+				_track[track].period -= exy;
+				break;
+			case 0x2: // Fine slide down
+				_track[track].period += exy;
+				break;
+			case 0x5: // Set finetune
+				_track[track].finetune = ey;
+				_module.sample[_track[track].sample].finetune = ey;
+				if (note.period) {
+					if (ey)
+						_track[track].period = _module.noteToPeriod(note.note, ey);
+					else
+						_track[track].period = note.period;
+				}
+				break;
+			case 0x6:
+				if (ey == 0) {
+					_patternLoopRow = _row;
+				} else {
+					_patternLoopCount++;
+					if (_patternLoopCount <= ey)
+						_hasPatternLoop = true;
+					else
+						_patternLoopCount = 0;
+				}
+				break;
+			case 0x9:
+				break;	// Retrigger note
+			case 0xA: // Fine volume slide up
+				vol = _track[track].vol + ey;
+				if (vol > 64)
+					vol = 64;
+				_track[track].vol = vol;
+				break;
+			case 0xB: // Fine volume slide down
+				vol = _track[track].vol - ey;
+				if (vol < 0)
+					vol = 0;
+				_track[track].vol = vol;
+				break;
+			case 0xD: // Delay sample
+				_track[track].delaySampleTick = ey;
+				_track[track].delaySample = _track[track].sample;
+				_track[track].sample = 0;
+				_track[track].vol = 0;
+				break;
+			case 0xE: // Pattern delay
+				_patternDelay = ey;
+				break;
+			default:
+				warning("Unimplemented effect %X", note.effect);
+			}
+			break;
+
+		case 0xF:
+			if (exy < 0x20) {
+				_speed = exy;
+			} else {
+				_bpm = exy;
+				setInterruptFreq((int) (getRate() / (_bpm * 0.4)));
+			}
+			break;
+		default:
+			warning("Unimplemented effect %X", note.effect);
+		}
+	}
+}
+
+void ProtrackerStream::updateEffects() {
+	for (int track = 0; track < 4; track++) {
+		_track[track].vibrato = 0;
+
+		const note_t note =
+		    _module.pattern[_module.songpos[_pos]][_row][track];
+
+		const int effect = note.effect >> 8;
+
+		const int exy = note.effect & 0xff;
+		const int ex = (note.effect >> 4) & 0xf;
+		const int ey = (note.effect) & 0xf;
+
+		switch (effect) {
+		case 0x0:
+			if (exy) {
+				const int idx = (_tick == 1) ? 0 : (_tick % 3);
+				_track[track].period =
+					_module.noteToPeriod(_track[track].arpeggioNotes[idx],
+							_track[track].finetune);
+			}
+			break;
+		case 0x1:
+			_track[track].period -= exy;
+			break;
+		case 0x2:
+			_track[track].period += exy;
+			break;
+		case 0x3:
+			doPorta(track);
+			break;
+		case 0x4:
+			doVibrato(track);
+			break;
+		case 0x5:
+			doPorta(track);
+			doVolSlide(track, ex, ey);
+			break;
+		case 0x6:
+			doVibrato(track);
+			doVolSlide(track, ex, ey);
+			break;
+		case 0xA:
+			doVolSlide(track, ex, ey);
+			break;
+		case 0xE:
+			switch (ex) {
+			case 0x6:
+				break;	// Pattern loop
+			case 0x9:	// Retrigger note
+				if (ey && (_tick % ey) == 0)
+					_track[track].offset = Offset(0);
+				break;
+			case 0xD: // Delay sample
+				if (_tick == _track[track].delaySampleTick) {
+					_track[track].sample = _track[track].delaySample;
+					_track[track].offset = Offset(0);
+					if (_track[track].sample)
+						_track[track].vol = _module.sample[_track[track].sample - 1].vol;
+				}
+				break;
+			}
+			break;
+		}
+	}
+}
+
+void ProtrackerStream::interrupt() {
+	int track;
+
+	for (track = 0; track < 4; track++) {
+		_track[track].offset = getChannelOffset(track);
+		if (_tick == 0 && _track[track].arpeggio) {
+			_track[track].period = _module.noteToPeriod(_track[track].arpeggioNotes[0],
+					_track[track].finetune);
+		}
+	}
+
+	if (_tick == 0) {
+		if (_hasJumpToPattern) {
+			_hasJumpToPattern = false;
+			_pos = _jumpToPattern;
+			_row = 0;
+		} else if (_hasPatternBreak) {
+			_hasPatternBreak = false;
+			_row = _skipRow;
+			_pos = (_pos + 1) % _module.songlen;
+			_patternLoopRow = 0;
+		} else if (_hasPatternLoop) {
+			_hasPatternLoop = false;
+			_row = _patternLoopRow;
+		}
+		if (_row >= 64) {
+			_row = 0;
+			_pos = (_pos + 1) % _module.songlen;
+			_patternLoopRow = 0;
+		}
+
+		updateRow();
+	} else
+		updateEffects();
+
+	_tick = (_tick + 1) % (_speed + _patternDelay * _speed);
+	if (_tick == 0) {
+		_row++;
+		_patternDelay = 0;
+	}
+
+	for (track = 0; track < 4; track++) {
+		setChannelVolume(track, _track[track].vol);
+		setChannelPeriod(track, _track[track].period + _track[track].vibrato);
+		if (_track[track].sample) {
+			sample_t &sample = _module.sample[_track[track].sample - 1];
+			setChannelData(track,
+			               sample.data,
+			               sample.replen > 2 ? sample.data + sample.repeat : 0,
+			               sample.len,
+			               sample.replen);
+			setChannelOffset(track, _track[track].offset);
+			_track[track].sample = 0;
+		}
+	}
+}
+
+} // End of namespace Modules
+
+namespace Audio {
+
+AudioStream *makeProtrackerStream(Common::SeekableReadStream *stream, int offs, int rate, bool stereo) {
+	return new Modules::ProtrackerStream(stream, offs, rate, stereo);
+}
+
+} // End of namespace Audio
diff --git a/audio/mods/protracker.h b/audio/mods/protracker.h
new file mode 100644
index 0000000000..af722637c7
--- /dev/null
+++ b/audio/mods/protracker.h
@@ -0,0 +1,57 @@
+/* ScummVM - Graphic Adventure Engine
+ *
+ * ScummVM is the legal property of its developers, whose names
+ * are too numerous to list here. Please refer to the COPYRIGHT
+ * file distributed with this source distribution.
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
+ *
+ * $URL$
+ * $Id$
+ *
+ */
+
+/**
+ * @file
+ * Sound decoder used in engines:
+ *  - agos
+ *  - parallaction
+ */
+
+#ifndef SOUND_MODS_PROTRACKER_H
+#define SOUND_MODS_PROTRACKER_H
+
+#include "common/stream.h"
+
+namespace Audio {
+
+class AudioStream;
+
+/*
+ * Factory function for ProTracker streams. Reads all data from the
+ * given ReadStream and creates an AudioStream from this. No reference
+ * to the 'stream' object is kept, so you can safely delete it after
+ * invoking this factory.
+ *
+ * @param stream	the ReadStream from which to read the ProTracker data
+ * @param rate		TODO
+ * @param stereo	TODO
+ * @return	a new AudioStream, or NULL, if an error occurred
+ */
+AudioStream *makeProtrackerStream(Common::SeekableReadStream *stream, int offs = 0, int rate = 44100, bool stereo = true);
+
+} // End of namespace Audio
+
+#endif
diff --git a/audio/mods/rjp1.cpp b/audio/mods/rjp1.cpp
new file mode 100644
index 0000000000..7423abb668
--- /dev/null
+++ b/audio/mods/rjp1.cpp
@@ -0,0 +1,582 @@
+/* ScummVM - Graphic Adventure Engine
+ *
+ * ScummVM is the legal property of its developers, whose names
+ * are too numerous to list here. Please refer to the COPYRIGHT
+ * file distributed with this source distribution.
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
+ *
+ * $URL$
+ * $Id$
+ *
+ */
+
+#include "common/debug.h"
+#include "common/endian.h"
+
+#include "audio/mods/paula.h"
+#include "audio/mods/rjp1.h"
+#include "audio/audiostream.h"
+
+namespace Audio {
+
+struct Rjp1Channel {
+	const int8 *waveData;
+	const int8 *modulatePeriodData;
+	const int8 *modulateVolumeData;
+	const int8 *envelopeData;
+	uint16 volumeScale;
+	int16 volume;
+	uint16 modulatePeriodBase;
+	uint32 modulatePeriodLimit;
+	uint32 modulatePeriodIndex;
+	uint16 modulateVolumeBase;
+	uint32 modulateVolumeLimit;
+	uint32 modulateVolumeIndex;
+	uint8 freqStep;
+	uint32 freqInc;
+	uint32 freqInit;
+	const uint8 *noteData;
+	const uint8 *sequenceOffsets;
+	const uint8 *sequenceData;
+	uint8 loopSeqCount;
+	uint8 loopSeqCur;
+	uint8 loopSeq2Count;
+	uint8 loopSeq2Cur;
+	bool active;
+	int16 modulatePeriodInit;
+	int16 modulatePeriodNext;
+	bool setupNewNote;
+	int8 envelopeMode;
+	int8 envelopeScale;
+	int8 envelopeEnd1;
+	int8 envelopeEnd2;
+	int8 envelopeStart;
+	int8 envelopeVolume;
+	uint8 currentInstrument;
+	const int8 *data;
+	uint16 pos;
+	uint16 len;
+	uint16 repeatPos;
+	uint16 repeatLen;
+	bool isSfx;
+};
+
+class Rjp1 : public Paula {
+public:
+
+	struct Vars {
+		int8 *instData;
+		uint8 *songData[7];
+		uint8 activeChannelsMask;
+		uint8 currentChannel;
+		int subsongsCount;
+		int instrumentsCount;
+	};
+
+	Rjp1(int rate, bool stereo);
+	virtual ~Rjp1();
+
+	bool load(Common::SeekableReadStream *songData, Common::SeekableReadStream *instrumentsData);
+	void unload();
+
+	void startPattern(int ch, int pat);
+	void startSong(int song);
+
+protected:
+
+	void startSequence(uint8 channelNum, uint8 seqNum);
+	void turnOffChannel(Rjp1Channel *channel);
+	void playChannel(Rjp1Channel *channel);
+	void turnOnChannel(Rjp1Channel *channel);
+	bool executeSfxSequenceOp(Rjp1Channel *channel, uint8 code, const uint8 *&p);
+	bool executeSongSequenceOp(Rjp1Channel *channel, uint8 code, const uint8 *&p);
+	void playSongSequence(Rjp1Channel *channel);
+	void modulateVolume(Rjp1Channel *channel);
+	void modulatePeriod(Rjp1Channel *channel);
+	void setupNote(Rjp1Channel *channel, int16 freq);
+	void setupInstrument(Rjp1Channel *channel, uint8 num);
+	void setRelease(Rjp1Channel *channel);
+	void modulateVolumeEnvelope(Rjp1Channel *channel);
+	void setSustain(Rjp1Channel *channel);
+	void setDecay(Rjp1Channel *channel);
+	void modulateVolumeWaveform(Rjp1Channel *channel);
+	void setVolume(Rjp1Channel *channel);
+
+	void stopPaulaChannel(uint8 channel);
+	void setupPaulaChannel(uint8 channel, const int8 *waveData, uint16 offset, uint16 len, uint16 repeatPos, uint16 repeatLen);
+
+	virtual void interrupt();
+
+	Vars _vars;
+	Rjp1Channel _channelsTable[4];
+
+	static const int16 _periodsTable[];
+	static const int _periodsCount;
+};
+
+Rjp1::Rjp1(int rate, bool stereo)
+	: Paula(stereo, rate, rate / 50) {
+	memset(&_vars, 0, sizeof(_vars));
+	memset(_channelsTable, 0, sizeof(_channelsTable));
+}
+
+Rjp1::~Rjp1() {
+	unload();
+}
+
+bool Rjp1::load(Common::SeekableReadStream *songData, Common::SeekableReadStream *instrumentsData) {
+	if (songData->readUint32BE() == MKID_BE('RJP1') && songData->readUint32BE() == MKID_BE('SMOD')) {
+		for (int i = 0; i < 7; ++i) {
+			uint32 size = songData->readUint32BE();
+			_vars.songData[i] = (uint8 *)malloc(size);
+			if (!_vars.songData[i])
+				return false;
+
+			songData->read(_vars.songData[i], size);
+			switch (i) {
+			case 0:
+				_vars.instrumentsCount = size / 32;
+				break;
+			case 1:
+				break;
+			case 2:
+				// sequence index to offsets, 1 per channel
+				_vars.subsongsCount = size / 4;
+				break;
+			case 3:
+			case 4:
+				// sequence offsets
+				break;
+			case 5:
+			case 6:
+				// sequence data
+				break;
+			}
+		}
+
+		if (instrumentsData->readUint32BE() == MKID_BE('RJP1')) {
+			uint32 size = instrumentsData->size() - 4;
+			_vars.instData = (int8 *)malloc(size);
+			if (!_vars.instData)
+				return false;
+
+			instrumentsData->read(_vars.instData, size);
+
+		}
+	}
+
+	debug(5, "Rjp1::load() _instrumentsCount = %d _subsongsCount = %d", _vars.instrumentsCount, _vars.subsongsCount);
+	return true;
+}
+
+void Rjp1::unload() {
+	for (int i = 0; i < 7; ++i) {
+		free(_vars.songData[i]);
+	}
+	free(_vars.instData);
+	memset(&_vars, 0, sizeof(_vars));
+	memset(_channelsTable, 0, sizeof(_channelsTable));
+}
+
+void Rjp1::startPattern(int ch, int pat) {
+	Rjp1Channel *channel = &_channelsTable[ch];
+	_vars.activeChannelsMask |= 1 << ch;
+	channel->sequenceData = READ_BE_UINT32(_vars.songData[4] + pat * 4) + _vars.songData[6];
+	channel->loopSeqCount = 6;
+	channel->loopSeqCur = channel->loopSeq2Cur = 1;
+	channel->active = true;
+	channel->isSfx = true;
+	// "start" Paula audiostream
+	startPaula();
+}
+
+void Rjp1::startSong(int song) {
+	if (song == 0 || song >= _vars.subsongsCount) {
+		warning("Invalid subsong number %d, defaulting to 1", song);
+		song = 1;
+	}
+	const uint8 *p = _vars.songData[2] + (song & 0x3F) * 4;
+	for (int i = 0; i < 4; ++i) {
+		uint8 seq = *p++;
+		if (seq) {
+			startSequence(i, seq);
+		}
+	}
+	// "start" Paula audiostream
+	startPaula();
+}
+
+void Rjp1::startSequence(uint8 channelNum, uint8 seqNum) {
+	Rjp1Channel *channel = &_channelsTable[channelNum];
+	_vars.activeChannelsMask |= 1 << channelNum;
+	if (seqNum != 0) {
+		const uint8 *p = READ_BE_UINT32(_vars.songData[3] + seqNum * 4) + _vars.songData[5];
+		uint8 seq = *p++;
+		channel->sequenceOffsets = p;
+		channel->sequenceData = READ_BE_UINT32(_vars.songData[4] + seq * 4) + _vars.songData[6];
+		channel->loopSeqCount = 6;
+		channel->loopSeqCur = channel->loopSeq2Cur = 1;
+		channel->active = true;
+	} else {
+		channel->active = false;
+		turnOffChannel(channel);
+	}
+}
+
+void Rjp1::turnOffChannel(Rjp1Channel *channel) {
+	stopPaulaChannel(channel - _channelsTable);
+}
+
+void Rjp1::playChannel(Rjp1Channel *channel) {
+	if (channel->active) {
+		turnOnChannel(channel);
+		if (channel->sequenceData) {
+			playSongSequence(channel);
+		}
+		modulateVolume(channel);
+		modulatePeriod(channel);
+	}
+}
+
+void Rjp1::turnOnChannel(Rjp1Channel *channel) {
+	if (channel->setupNewNote) {
+		channel->setupNewNote = false;
+		setupPaulaChannel(channel - _channelsTable, channel->data, channel->pos, channel->len, channel->repeatPos, channel->repeatLen);
+	}
+}
+
+bool Rjp1::executeSfxSequenceOp(Rjp1Channel *channel, uint8 code, const uint8 *&p) {
+	bool loop = true;
+	switch (code & 7) {
+	case 0:
+		_vars.activeChannelsMask &= ~(1 << _vars.currentChannel);
+		loop = false;
+		stopPaula();
+		break;
+	case 1:
+		setRelease(channel);
+		loop = false;
+		break;
+	case 2:
+		channel->loopSeqCount = *p++;
+		break;
+	case 3:
+		channel->loopSeq2Count = *p++;
+		break;
+	case 4:
+		code = *p++;
+		if (code != 0) {
+			setupInstrument(channel, code);
+		}
+		break;
+	case 7:
+		loop = false;
+		break;
+	}
+	return loop;
+}
+
+bool Rjp1::executeSongSequenceOp(Rjp1Channel *channel, uint8 code, const uint8 *&p) {
+	bool loop = true;
+	const uint8 *offs;
+	switch (code & 7) {
+	case 0:
+		offs = channel->sequenceOffsets;
+		channel->loopSeq2Count = 1;
+		while (1) {
+			code = *offs++;
+			if (code != 0) {
+				channel->sequenceOffsets = offs;
+				p = READ_BE_UINT32(_vars.songData[4] + code * 4) + _vars.songData[6];
+				break;
+			} else {
+				code = offs[0];
+				if (code == 0) {
+					p = 0;
+					channel->active = false;
+					_vars.activeChannelsMask &= ~(1 << _vars.currentChannel);
+					loop = false;
+					break;
+				} else if (code & 0x80) {
+					code = offs[1];
+					offs = READ_BE_UINT32(_vars.songData[3] + code * 4) + _vars.songData[5];
+				} else {
+					offs -= code;
+				}
+			}
+		}
+		break;
+	case 1:
+		setRelease(channel);
+		loop = false;
+		break;
+	case 2:
+		channel->loopSeqCount = *p++;
+		break;
+	case 3:
+		channel->loopSeq2Count = *p++;
+		break;
+	case 4:
+		code = *p++;
+		if (code != 0) {
+			setupInstrument(channel, code);
+		}
+		break;
+	case 5:
+		channel->volumeScale = *p++;
+		break;
+	case 6:
+		channel->freqStep = *p++;
+		channel->freqInc = READ_BE_UINT32(p); p += 4;
+		channel->freqInit = 0;
+		break;
+	case 7:
+		loop = false;
+		break;
+	}
+	return loop;
+}
+
+void Rjp1::playSongSequence(Rjp1Channel *channel) {
+	const uint8 *p = channel->sequenceData;
+	--channel->loopSeqCur;
+	if (channel->loopSeqCur == 0) {
+		--channel->loopSeq2Cur;
+		if (channel->loopSeq2Cur == 0) {
+			bool loop = true;
+			do {
+				uint8 code = *p++;
+				if (code & 0x80) {
+					if (channel->isSfx) {
+						loop = executeSfxSequenceOp(channel, code, p);
+					} else {
+						loop = executeSongSequenceOp(channel, code, p);
+					}
+				} else {
+					code >>= 1;
+					if (code < _periodsCount) {
+						setupNote(channel, _periodsTable[code]);
+					}
+					loop = false;
+				}
+			} while (loop);
+			channel->sequenceData = p;
+			channel->loopSeq2Cur = channel->loopSeq2Count;
+		}
+		channel->loopSeqCur = channel->loopSeqCount;
+	}
+}
+
+void Rjp1::modulateVolume(Rjp1Channel *channel) {
+	modulateVolumeEnvelope(channel);
+	modulateVolumeWaveform(channel);
+	setVolume(channel);
+}
+
+void Rjp1::modulatePeriod(Rjp1Channel *channel) {
+	if (channel->modulatePeriodData) {
+		uint32 per = channel->modulatePeriodIndex;
+		int period = (channel->modulatePeriodData[per] * channel->modulatePeriodInit) / 128;
+		period = -period;
+		if (period < 0) {
+			period /= 2;
+		}
+		channel->modulatePeriodNext = period + channel->modulatePeriodInit;
+		++per;
+		if (per == channel->modulatePeriodLimit) {
+			per = channel->modulatePeriodBase * 2;
+		}
+		channel->modulatePeriodIndex = per;
+	}
+	if (channel->freqStep != 0) {
+		channel->freqInit += channel->freqInc;
+		--channel->freqStep;
+	}
+	setChannelPeriod(channel - _channelsTable, channel->freqInit + channel->modulatePeriodNext);
+}
+
+void Rjp1::setupNote(Rjp1Channel *channel, int16 period) {
+	const uint8 *note = channel->noteData;
+	if (note) {
+		channel->modulatePeriodInit = channel->modulatePeriodNext = period;
+		channel->freqInit = 0;
+		const int8 *e = (const int8 *)_vars.songData[1] + READ_BE_UINT16(note + 12);
+		channel->envelopeData = e;
+		channel->envelopeStart = e[1];
+		channel->envelopeScale = e[1] - e[0];
+		channel->envelopeEnd2 = e[2];
+		channel->envelopeEnd1 = e[2];
+		channel->envelopeMode = 4;
+		channel->data = channel->waveData;
+		channel->pos = READ_BE_UINT16(note + 16);
+		channel->len = channel->pos + READ_BE_UINT16(note + 18);
+		channel->setupNewNote = true;
+	}
+}
+
+void Rjp1::setupInstrument(Rjp1Channel *channel, uint8 num) {
+	if (channel->currentInstrument != num) {
+		channel->currentInstrument = num;
+		const uint8 *p = _vars.songData[0] + num * 32;
+		channel->noteData = p;
+		channel->repeatPos = READ_BE_UINT16(p + 20);
+		channel->repeatLen = READ_BE_UINT16(p + 22);
+		channel->volumeScale = READ_BE_UINT16(p + 14);
+		channel->modulatePeriodBase = READ_BE_UINT16(p + 24);
+		channel->modulatePeriodIndex = 0;
+		channel->modulatePeriodLimit = READ_BE_UINT16(p + 26) * 2;
+		channel->modulateVolumeBase = READ_BE_UINT16(p + 28);
+		channel->modulateVolumeIndex = 0;
+		channel->modulateVolumeLimit = READ_BE_UINT16(p + 30) * 2;
+		channel->waveData = _vars.instData + READ_BE_UINT32(p);
+		uint32 off = READ_BE_UINT32(p + 4);
+		if (off) {
+			channel->modulatePeriodData = _vars.instData + off;
+		}
+		off = READ_BE_UINT32(p + 8);
+		if (off) {
+			channel->modulateVolumeData = _vars.instData + off;
+		}
+	}
+}
+
+void Rjp1::setRelease(Rjp1Channel *channel) {
+	const int8 *e = channel->envelopeData;
+	if (e) {
+		channel->envelopeStart = 0;
+		channel->envelopeScale = -channel->envelopeVolume;
+		channel->envelopeEnd2 = e[5];
+		channel->envelopeEnd1 = e[5];
+		channel->envelopeMode = -1;
+	}
+}
+
+void Rjp1::modulateVolumeEnvelope(Rjp1Channel *channel) {
+	if (channel->envelopeMode) {
+		int16 es = channel->envelopeScale;
+		if (es) {
+			int8 m = channel->envelopeEnd1;
+			if (m == 0) {
+				es = 0;
+			} else {
+				es *= m;
+				m = channel->envelopeEnd2;
+				if (m == 0) {
+					es = 0;
+				} else {
+					es /= m;
+				}
+			}
+		}
+		channel->envelopeVolume = channel->envelopeStart - es;
+		--channel->envelopeEnd1;
+		if (channel->envelopeEnd1 == -1) {
+			switch (channel->envelopeMode) {
+			case 0:
+				break;
+			case 2:
+				setSustain(channel);
+				break;
+			case 4:
+				setDecay(channel);
+				break;
+			case -1:
+				setSustain(channel);
+				break;
+			default:
+				error("Unhandled envelope mode %d", channel->envelopeMode);
+				break;
+			}
+			return;
+		}
+	}
+	channel->volume = channel->envelopeVolume;
+}
+
+void Rjp1::setSustain(Rjp1Channel *channel) {
+	channel->envelopeMode = 0;
+}
+
+void Rjp1::setDecay(Rjp1Channel *channel) {
+	const int8 *e = channel->envelopeData;
+	if (e) {
+		channel->envelopeStart = e[3];
+		channel->envelopeScale = e[3] - e[1];
+		channel->envelopeEnd2 = e[4];
+		channel->envelopeEnd1 = e[4];
+		channel->envelopeMode = 2;
+	}
+}
+
+void Rjp1::modulateVolumeWaveform(Rjp1Channel *channel) {
+	if (channel->modulateVolumeData) {
+		uint32 i = channel->modulateVolumeIndex;
+		channel->volume += channel->modulateVolumeData[i] * channel->volume / 128;
+		++i;
+		if (i == channel->modulateVolumeLimit) {
+			i = channel->modulateVolumeBase * 2;
+		}
+		channel->modulateVolumeIndex = i;
+	}
+}
+
+void Rjp1::setVolume(Rjp1Channel *channel) {
+	channel->volume = (channel->volume * channel->volumeScale) / 64;
+	channel->volume = CLIP<int16>(channel->volume, 0, 64);
+	setChannelVolume(channel - _channelsTable, channel->volume);
+}
+
+void Rjp1::stopPaulaChannel(uint8 channel) {
+	clearVoice(channel);
+}
+
+void Rjp1::setupPaulaChannel(uint8 channel, const int8 *waveData, uint16 offset, uint16 len, uint16 repeatPos, uint16 repeatLen) {
+	if (waveData) {
+		setChannelData(channel, waveData, waveData + repeatPos * 2, len * 2, repeatLen * 2, offset * 2);
+	}
+}
+
+void Rjp1::interrupt() {
+	for (int i = 0; i < 4; ++i) {
+		_vars.currentChannel = i;
+		playChannel(&_channelsTable[i]);
+	}
+}
+
+const int16 Rjp1::_periodsTable[] = {
+	0x01C5, 0x01E0, 0x01FC, 0x021A, 0x023A, 0x025C, 0x0280, 0x02A6, 0x02D0,
+	0x02FA, 0x0328, 0x0358, 0x00E2, 0x00F0, 0x00FE, 0x010D, 0x011D, 0x012E,
+	0x0140, 0x0153, 0x0168, 0x017D, 0x0194, 0x01AC, 0x0071, 0x0078, 0x007F,
+	0x0087, 0x008F, 0x0097, 0x00A0, 0x00AA, 0x00B4, 0x00BE, 0x00CA, 0x00D6
+};
+
+const int Rjp1::_periodsCount = ARRAYSIZE(_periodsTable);
+
+AudioStream *makeRjp1Stream(Common::SeekableReadStream *songData, Common::SeekableReadStream *instrumentsData, int num, int rate, bool stereo) {
+	Rjp1 *stream = new Rjp1(rate, stereo);
+	if (stream->load(songData, instrumentsData)) {
+		if (num < 0) {
+			stream->startPattern(3, -num);
+		} else {
+			stream->startSong(num);
+		}
+		return stream;
+	}
+	delete stream;
+	return 0;
+}
+
+} // End of namespace Audio
diff --git a/audio/mods/rjp1.h b/audio/mods/rjp1.h
new file mode 100644
index 0000000000..e1960921b2
--- /dev/null
+++ b/audio/mods/rjp1.h
@@ -0,0 +1,50 @@
+/* ScummVM - Graphic Adventure Engine
+ *
+ * ScummVM is the legal property of its developers, whose names
+ * are too numerous to list here. Please refer to the COPYRIGHT
+ * file distributed with this source distribution.
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
+ *
+ * $URL$
+ * $Id$
+ *
+ */
+
+/**
+ * @file
+ * Sound decoder used in engines:
+ *  - queen
+ */
+
+#ifndef SOUND_MODS_RJP1_H
+#define SOUND_MODS_RJP1_H
+
+#include "common/stream.h"
+
+namespace Audio {
+
+class AudioStream;
+
+/*
+ * Factory function for RichardJoseph1 modules. Reads all data from the
+ * given songData and instrumentsData streams and creates an AudioStream
+ * from this. No references to these stream objects are kept.
+ */
+AudioStream *makeRjp1Stream(Common::SeekableReadStream *songData, Common::SeekableReadStream *instrumentsData, int num, int rate = 44100, bool stereo = true);
+
+} // End of namespace Audio
+
+#endif
diff --git a/audio/mods/soundfx.cpp b/audio/mods/soundfx.cpp
new file mode 100644
index 0000000000..06a1e29514
--- /dev/null
+++ b/audio/mods/soundfx.cpp
@@ -0,0 +1,275 @@
+/* ScummVM - Graphic Adventure Engine
+ *
+ * ScummVM is the legal property of its developers, whose names
+ * are too numerous to list here. Please refer to the COPYRIGHT
+ * file distributed with this source distribution.
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
+ *
+ * $URL$
+ * $Id$
+ *
+ */
+
+#include "common/endian.h"
+
+#include "audio/mods/paula.h"
+#include "audio/mods/soundfx.h"
+#include "audio/audiostream.h"
+
+namespace Audio {
+
+struct SoundFxInstrument {
+	char name[23];
+	uint16 len;
+	uint8 finetune;
+	uint8 volume;
+	uint16 repeatPos;
+	uint16 repeatLen;
+	int8 *data;
+};
+
+class SoundFx : public Paula {
+public:
+
+	enum {
+		NUM_CHANNELS = 4,
+		NUM_INSTRUMENTS = 15
+	};
+
+	SoundFx(int rate, bool stereo);
+	virtual ~SoundFx();
+
+	bool load(Common::SeekableReadStream *data, LoadSoundFxInstrumentCallback loadCb);
+	void play();
+
+protected:
+
+	void handlePattern(int ch, uint32 pat);
+	void updateEffects(int ch);
+	void handleTick();
+
+	void disablePaulaChannel(uint8 channel);
+	void setupPaulaChannel(uint8 channel, const int8 *data, uint16 len, uint16 repeatPos, uint16 repeatLen);
+
+	virtual void interrupt();
+
+	uint8 _ticks;
+	uint16 _delay;
+	SoundFxInstrument _instruments[NUM_INSTRUMENTS];
+	uint8 _numOrders;
+	uint8 _curOrder;
+	uint16 _curPos;
+	uint8 _ordersTable[128];
+	uint8 *_patternData;
+	uint16 _effects[NUM_CHANNELS];
+};
+
+SoundFx::SoundFx(int rate, bool stereo)
+	: Paula(stereo, rate) {
+	setTimerBaseValue(kPalCiaClock);
+	_ticks = 0;
+	_delay = 0;
+	memset(_instruments, 0, sizeof(_instruments));
+	_numOrders = 0;
+	_curOrder = 0;
+	_curPos = 0;
+	memset(_ordersTable, 0, sizeof(_ordersTable));
+	_patternData = 0;
+	memset(_effects, 0, sizeof(_effects));
+}
+
+SoundFx::~SoundFx() {
+	free(_patternData);
+	for (int i = 0; i < NUM_INSTRUMENTS; ++i) {
+		free(_instruments[i].data);
+	}
+}
+
+bool SoundFx::load(Common::SeekableReadStream *data, LoadSoundFxInstrumentCallback loadCb) {
+	int instrumentsSize[15];
+	if (!loadCb) {
+		for (int i = 0; i < NUM_INSTRUMENTS; ++i) {
+			instrumentsSize[i] = data->readUint32BE();
+		}
+	}
+	uint8 tag[4];
+	data->read(tag, 4);
+	if (memcmp(tag, "SONG", 4) != 0) {
+		return false;
+	}
+	_delay = data->readUint16BE();
+	data->skip(7 * 2);
+	for (int i = 0; i < NUM_INSTRUMENTS; ++i) {
+		SoundFxInstrument *ins = &_instruments[i];
+		data->read(ins->name, 22); ins->name[22] = 0;
+		ins->len = data->readUint16BE();
+		ins->finetune = data->readByte();
+		ins->volume = data->readByte();
+		ins->repeatPos = data->readUint16BE();
+		ins->repeatLen = data->readUint16BE();
+	}
+	_numOrders = data->readByte();
+	data->skip(1);
+	data->read(_ordersTable, 128);
+	int maxOrder = 0;
+	for (int i = 0; i < _numOrders; ++i) {
+		if (_ordersTable[i] > maxOrder) {
+			maxOrder = _ordersTable[i];
+		}
+	}
+	int patternSize = (maxOrder + 1) * 4 * 4 * 64;
+	_patternData = (uint8 *)malloc(patternSize);
+	if (!_patternData) {
+		return false;
+	}
+	data->read(_patternData, patternSize);
+	for (int i = 0; i < NUM_INSTRUMENTS; ++i) {
+		SoundFxInstrument *ins = &_instruments[i];
+		if (!loadCb) {
+			if (instrumentsSize[i] != 0) {
+				assert(ins->len <= 1 || ins->len * 2 <= instrumentsSize[i]);
+				assert(ins->repeatLen <= 1 || (ins->repeatPos + ins->repeatLen) * 2 <= instrumentsSize[i]);
+				ins->data = (int8 *)malloc(instrumentsSize[i]);
+				if (!ins->data) {
+					return false;
+				}
+				data->read(ins->data, instrumentsSize[i]);
+			}
+		} else {
+			if (ins->name[0]) {
+				ins->name[8] = '\0';
+				ins->data = (int8 *)(*loadCb)(ins->name, 0);
+				if (!ins->data) {
+					return false;
+				}
+			}
+		}
+	}
+	return true;
+}
+
+void SoundFx::play() {
+	_curPos = 0;
+	_curOrder = 0;
+	_ticks = 0;
+	setInterruptFreqUnscaled(_delay);
+	startPaula();
+}
+
+void SoundFx::handlePattern(int ch, uint32 pat) {
+	uint16 note1 = pat >> 16;
+	uint16 note2 = pat & 0xFFFF;
+	if (note1 == 0xFFFD) { // PIC
+		_effects[ch] = 0;
+		return;
+	}
+	_effects[ch] = note2;
+	if (note1 == 0xFFFE) { // STP
+		disablePaulaChannel(ch);
+		return;
+	}
+	int ins = (note2 & 0xF000) >> 12;
+	if (ins != 0) {
+		SoundFxInstrument *i = &_instruments[ins - 1];
+		setupPaulaChannel(ch, i->data, i->len, i->repeatPos, i->repeatLen);
+		int effect = (note2 & 0xF00) >> 8;
+		int volume = i->volume;
+		switch (effect) {
+		case 5: // volume up
+			volume += (note2 & 0xFF);
+			if (volume > 63) {
+				volume = 63;
+			}
+			break;
+		case 6: // volume down
+			volume -= (note2 & 0xFF);
+			if (volume < 0) {
+				volume = 0;
+			}
+			break;
+		}
+		setChannelVolume(ch, volume);
+	}
+	if (note1 != 0) {
+		setChannelPeriod(ch, note1);
+	}
+}
+
+void SoundFx::updateEffects(int ch) {
+	// updateEffects() is a no-op in all Delphine Software games using SoundFx : FW,OS,Cruise,AW
+	if (_effects[ch] != 0) {
+		switch (_effects[ch]) {
+		case 1: // appreggiato
+		case 2: // pitchbend
+		case 3: // ledon, enable low-pass filter
+		case 4: // ledoff, disable low-pass filter
+		case 7: // set step up
+		case 8: // set step down
+			warning("Unhandled effect %d", _effects[ch]);
+			break;
+		}
+	}
+}
+
+void SoundFx::handleTick() {
+	++_ticks;
+	if (_ticks != 6) {
+		for (int ch = 0; ch < 4; ++ch) {
+			updateEffects(ch);
+		}
+	} else {
+		_ticks = 0;
+		const uint8 *patternData = _patternData + _ordersTable[_curOrder] * 1024 + _curPos;
+		for (int ch = 0; ch < 4; ++ch) {
+			handlePattern(ch, READ_BE_UINT32(patternData));
+			patternData += 4;
+		}
+		_curPos += 4 * 4;
+		if (_curPos >= 1024) {
+			_curPos = 0;
+			++_curOrder;
+			if (_curOrder == _numOrders) {
+				stopPaula();
+			}
+		}
+	}
+}
+
+void SoundFx::disablePaulaChannel(uint8 channel) {
+	disableChannel(channel);
+}
+
+void SoundFx::setupPaulaChannel(uint8 channel, const int8 *data, uint16 len, uint16 repeatPos, uint16 repeatLen) {
+	if (data && len > 1) {
+		setChannelData(channel, data, data + repeatPos * 2, len * 2, repeatLen * 2);
+	}
+}
+
+void SoundFx::interrupt() {
+	handleTick();
+}
+
+AudioStream *makeSoundFxStream(Common::SeekableReadStream *data, LoadSoundFxInstrumentCallback loadCb, int rate, bool stereo) {
+	SoundFx *stream = new SoundFx(rate, stereo);
+	if (stream->load(data, loadCb)) {
+		stream->play();
+		return stream;
+	}
+	delete stream;
+	return 0;
+}
+
+} // End of namespace Audio
diff --git a/audio/mods/soundfx.h b/audio/mods/soundfx.h
new file mode 100644
index 0000000000..089c19d292
--- /dev/null
+++ b/audio/mods/soundfx.h
@@ -0,0 +1,53 @@
+/* ScummVM - Graphic Adventure Engine
+ *
+ * ScummVM is the legal property of its developers, whose names
+ * are too numerous to list here. Please refer to the COPYRIGHT
+ * file distributed with this source distribution.
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
+ *
+ * $URL$
+ * $Id$
+ *
+ */
+
+/**
+ * @file
+ * Sound decoder used in engines:
+ *  - cine
+ */
+
+#ifndef SOUND_MODS_SOUNDFX_H
+#define SOUND_MODS_SOUNDFX_H
+
+#include "common/stream.h"
+
+namespace Audio {
+
+class AudioStream;
+
+typedef byte *(*LoadSoundFxInstrumentCallback)(const char *name, uint32 *size);
+
+/*
+ * Factory function for SoundFX modules. Reads all data from the
+ * given data stream and creates an AudioStream from this (no references to the
+ * stream object is kept). If loadCb is non 0, then instruments are loaded using
+ * it, buffers returned are free'd at the end of playback.
+ */
+AudioStream *makeSoundFxStream(Common::SeekableReadStream *data, LoadSoundFxInstrumentCallback loadCb, int rate = 44100, bool stereo = true);
+
+} // End of namespace Audio
+
+#endif
diff --git a/audio/mods/tfmx.cpp b/audio/mods/tfmx.cpp
new file mode 100644
index 0000000000..8c69a75ebd
--- /dev/null
+++ b/audio/mods/tfmx.cpp
@@ -0,0 +1,1193 @@
+/* ScummVM - Graphic Adventure Engine
+ *
+ * ScummVM is the legal property of its developers, whose names
+ * are too numerous to list here. Please refer to the COPYRIGHT
+ * file distributed with this source distribution.
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
+ *
+ * $URL$
+ * $Id$
+ *
+ */
+
+#include "common/scummsys.h"
+#include "common/endian.h"
+#include "common/stream.h"
+#include "common/util.h"
+#include "common/debug.h"
+
+#include "audio/mods/tfmx.h"
+
+// test for engines using this class.
+#if defined(SOUND_MODS_TFMX_H)
+
+// couple debug-functions
+namespace {
+
+#if 0
+void displayPatternstep(const void * const vptr);
+void displayMacroStep(const void * const vptr);
+#endif
+
+static const uint16 noteIntervalls[64] = {
+	1710, 1614, 1524, 1438, 1357, 1281, 1209, 1141, 1077, 1017,  960,  908,
+	 856,  810,  764,  720,  680,  642,  606,  571,  539,  509,  480,  454,
+	 428,  404,  381,  360,  340,  320,  303,  286,  270,  254,  240,  227,
+	 214,  202,  191,  180,  170,  160,  151,  143,  135,  127,  120,  113,
+	 214,  202,  191,  180,  170,  160,  151,  143,  135,  127,  120,  113,
+	 214,  202,  191,  180
+};
+
+} // End of anonymous namespace
+
+namespace Audio {
+
+Tfmx::Tfmx(int rate, bool stereo)
+	: Paula(stereo, rate),
+	  _resource(),
+	  _resourceSample(),
+	  _playerCtx(),
+	  _deleteResource(false) {
+
+	_playerCtx.stopWithLastPattern = false;
+
+	for (int i = 0; i < kNumVoices; ++i)
+		_channelCtx[i].paulaChannel = (byte)i;
+
+	_playerCtx.volume = 0x40;
+	_playerCtx.patternSkip = 6;
+	stopSongImpl();
+
+	setTimerBaseValue(kPalCiaClock);
+	setInterruptFreqUnscaled(kPalDefaultCiaVal);
+}
+
+Tfmx::~Tfmx() {
+	freeResourceDataImpl();
+}
+
+void Tfmx::interrupt() {
+	assert(!_end);
+	++_playerCtx.tickCount;
+
+	for (int i = 0; i < kNumVoices; ++i) {
+		if (_channelCtx[i].dmaIntCount) {
+			// wait for DMA Interupts to happen
+			int doneDma = getChannelDmaCount(i);
+			if (doneDma >= _channelCtx[i].dmaIntCount) {
+				_channelCtx[i].dmaIntCount = 0;
+				_channelCtx[i].macroRun = true;
+			}
+		}
+	}
+
+	for (int i = 0; i < kNumVoices; ++i) {
+		ChannelContext &channel = _channelCtx[i];
+
+		if (channel.sfxLockTime >= 0)
+			--channel.sfxLockTime;
+		else {
+			channel.sfxLocked = false;
+			channel.customMacroPrio = 0;
+		}
+
+		// externally queued macros
+		if (channel.customMacro) {
+			const byte * const noteCmd = (const byte *)&channel.customMacro;
+			channel.sfxLocked = false;
+			noteCommand(noteCmd[0], noteCmd[1], (noteCmd[2] & 0xF0) | (uint8)i, noteCmd[3]);
+			channel.customMacro = 0;
+			channel.sfxLocked = (channel.customMacroPrio != 0);
+		}
+
+		// apply timebased effects on Parameters
+		if (channel.macroSfxRun > 0)
+			effects(channel);
+
+		// see if we have to run the macro-program
+		if (channel.macroRun) {
+			if (!channel.macroWait)
+				macroRun(channel);
+			else
+				--channel.macroWait;
+		}
+
+		Paula::setChannelPeriod(i, channel.period);
+		if (channel.macroSfxRun >= 0)
+			channel.macroSfxRun = 1;
+
+		// TODO: handling pending DMAOff?
+	}
+
+	// Patterns are only processed each _playerCtx.timerCount + 1 tick
+	if (_playerCtx.song >= 0 && !_playerCtx.patternCount--) {
+		_playerCtx.patternCount = _playerCtx.patternSkip;
+		advancePatterns();
+	}
+}
+
+void Tfmx::effects(ChannelContext &channel) {
+	// addBegin
+	if (channel.addBeginLength) {
+		channel.sampleStart += channel.addBeginDelta;
+		Paula::setChannelSampleStart(channel.paulaChannel, getSamplePtr(channel.sampleStart));
+		if (!(--channel.addBeginCount)) {
+			channel.addBeginCount = channel.addBeginLength;
+			channel.addBeginDelta = -channel.addBeginDelta;
+		}
+	}
+
+	// vibrato
+	if (channel.vibLength) {
+		channel.vibValue += channel.vibDelta;
+		if (--channel.vibCount == 0) {
+			channel.vibCount = channel.vibLength;
+			channel.vibDelta = -channel.vibDelta;
+		}
+		if (!channel.portaDelta) {
+			// 16x16 bit multiplication, casts needed for the right results
+			channel.period = (uint16)(((uint32)channel.refPeriod * (uint16)((1 << 11) + channel.vibValue)) >> 11);
+		}
+	}
+
+	// portamento
+	if (channel.portaDelta && !(--channel.portaCount)) {
+		channel.portaCount = channel.portaSkip;
+
+		bool resetPorta = true;
+		const uint16 period = channel.refPeriod;
+		uint16 portaVal = channel.portaValue;
+
+		if (period > portaVal) {
+			portaVal = ((uint32)portaVal * (uint16)((1 << 8) + channel.portaDelta)) >> 8;
+			resetPorta = (period <= portaVal);
+
+		} else if (period < portaVal) {
+			portaVal = ((uint32)portaVal * (uint16)((1 << 8) - channel.portaDelta)) >> 8;
+			resetPorta = (period >= portaVal);
+		}
+
+		if (resetPorta) {
+			channel.portaDelta = 0;
+			channel.portaValue = period & 0x7FF;
+		} else
+			channel.period = channel.portaValue = portaVal & 0x7FF;
+	}
+
+	// envelope
+	if (channel.envSkip && !channel.envCount--) {
+		channel.envCount = channel.envSkip;
+
+		const int8 endVol = channel.envEndVolume;
+		int8 volume = channel.volume;
+		bool resetEnv = true;
+
+		if (endVol > volume) {
+			volume += channel.envDelta;
+			resetEnv = endVol <= volume;
+		} else {
+			volume -= channel.envDelta;
+			resetEnv = volume <= 0 || endVol >= volume;
+		}
+
+		if (resetEnv) {
+			channel.envSkip = 0;
+			volume = endVol;
+		}
+		channel.volume = volume;
+	}
+
+	// Fade
+	if (_playerCtx.fadeDelta && !(--_playerCtx.fadeCount)) {
+		_playerCtx.fadeCount = _playerCtx.fadeSkip;
+
+		_playerCtx.volume += _playerCtx.fadeDelta;
+		if (_playerCtx.volume == _playerCtx.fadeEndVolume)
+			_playerCtx.fadeDelta = 0;
+	}
+
+	// Volume
+	const uint8 finVol = _playerCtx.volume * channel.volume >> 6;
+	Paula::setChannelVolume(channel.paulaChannel, finVol);
+}
+
+void Tfmx::macroRun(ChannelContext &channel) {
+	bool deferWait = channel.deferWait;
+	for (;;) {
+		const byte *const macroPtr = (const byte *)(getMacroPtr(channel.macroOffset) + channel.macroStep);
+		++channel.macroStep;
+
+		switch (macroPtr[0]) {
+		case 0x00:	// Reset + DMA Off. Parameters: deferWait, addset, vol
+			clearEffects(channel);
+			// FT
+		case 0x13:	// DMA Off. Parameters:  deferWait, addset, vol
+			// TODO: implement PArameters
+			Paula::disableChannel(channel.paulaChannel);
+			channel.deferWait = deferWait = (macroPtr[1] != 0);
+			if (deferWait) {
+				// if set, then we expect a DMA On in the same tick.
+				channel.period = 4;
+				//Paula::setChannelPeriod(channel.paulaChannel, channel.period);
+				Paula::setChannelSampleLen(channel.paulaChannel, 1);
+				// in this state we then need to allow some commands that normally
+				// would halt the macroprogamm to continue instead.
+				// those commands are: Wait, WaitDMA, AddPrevNote, AddNote, SetNote, <unknown Cmd>
+				// DMA On is affected aswell
+				// TODO remember time disabled, remember pending dmaoff?.
+			}
+
+			if (macroPtr[2] || macroPtr[3]) {
+				channel.volume = (macroPtr[2] ? 0 : channel.relVol * 3) + macroPtr[3];
+				Paula::setChannelVolume(channel.paulaChannel, channel.volume);
+			}
+			continue;
+
+		case 0x01:	// DMA On
+			// TODO: Parameter macroPtr[1] - en-/disable effects
+			channel.dmaIntCount = 0;
+			if (deferWait) {
+				// TODO
+				// there is actually a small delay in the player, but I think that
+				// only allows to clear DMA-State on real Hardware
+			}
+			Paula::setChannelPeriod(channel.paulaChannel, channel.period);
+			Paula::enableChannel(channel.paulaChannel);
+			channel.deferWait = deferWait = false;
+			continue;
+
+		case 0x02:	// Set Beginn. Parameters: SampleOffset(L)
+			channel.addBeginLength = 0;
+			channel.sampleStart = READ_BE_UINT32(macroPtr) & 0xFFFFFF;
+			Paula::setChannelSampleStart(channel.paulaChannel, getSamplePtr(channel.sampleStart));
+			continue;
+
+		case 0x03:	// SetLength. Parameters: SampleLength(W)
+			channel.sampleLen = READ_BE_UINT16(&macroPtr[2]);
+			Paula::setChannelSampleLen(channel.paulaChannel, channel.sampleLen);
+			continue;
+
+		case 0x04:	// Wait. Parameters: Ticks to wait(W).
+			// TODO: some unknown Parameter? (macroPtr[1] & 1)
+			channel.macroWait = READ_BE_UINT16(&macroPtr[2]);
+			break;
+
+		case 0x10:	// Loop Key Up. Parameters: Loopcount, MacroStep(W)
+			if (channel.keyUp)
+				continue;
+			// FT
+		case 0x05:	// Loop. Parameters: Loopcount, MacroStep(W)
+			if (channel.macroLoopCount != 0) {
+				if (channel.macroLoopCount == 0xFF)
+					channel.macroLoopCount = macroPtr[1];
+				channel.macroStep = READ_BE_UINT16(&macroPtr[2]);
+			}
+			--channel.macroLoopCount;
+			continue;
+
+		case 0x06:	// Jump. Parameters: MacroIndex, MacroStep(W)
+			// channel.macroIndex = macroPtr[1] & (kMaxMacroOffsets - 1);
+			channel.macroOffset = _resource->macroOffset[macroPtr[1] & (kMaxMacroOffsets - 1)];
+			channel.macroStep = READ_BE_UINT16(&macroPtr[2]);
+			channel.macroLoopCount = 0xFF;
+			continue;
+
+		case 0x07:	// Stop Macro
+			channel.macroRun = false;
+			--channel.macroStep;
+			return;
+
+		case 0x08:	// AddNote. Parameters: Note, Finetune(W)
+			setNoteMacro(channel, channel.note + macroPtr[1], READ_BE_UINT16(&macroPtr[2]));
+			break;
+
+		case 0x09:	// SetNote. Parameters: Note, Finetune(W)
+			setNoteMacro(channel, macroPtr[1], READ_BE_UINT16(&macroPtr[2]));
+			break;
+
+		case 0x0A:	// Clear Effects
+			clearEffects(channel);
+			continue;
+
+		case 0x0B:	// Portamento. Parameters: count, speed
+			channel.portaSkip = macroPtr[1];
+			channel.portaCount = 1;
+			// if porta is already running, then keep using old value
+			if (!channel.portaDelta)
+				channel.portaValue = channel.refPeriod;
+			channel.portaDelta = READ_BE_UINT16(&macroPtr[2]);
+			continue;
+
+		case 0x0C:	// Vibrato. Parameters: Speed, intensity
+			channel.vibLength = macroPtr[1];
+			channel.vibCount = macroPtr[1] / 2;
+			channel.vibDelta = macroPtr[3];
+			// TODO: Perhaps a bug, vibValue could be left uninitialised
+			if (!channel.portaDelta) {
+				channel.period = channel.refPeriod;
+				channel.vibValue = 0;
+			}
+			continue;
+
+		case 0x0D:	// Add Volume. Parameters: note, addNoteFlag, volume
+			if (macroPtr[2] == 0xFE)
+				setNoteMacro(channel, channel.note + macroPtr[1], 0);
+			channel.volume = channel.relVol * 3 + macroPtr[3];
+			continue;
+
+		case 0x0E:	// Set Volume. Parameters: note, addNoteFlag, volume
+			if (macroPtr[2] == 0xFE)
+				setNoteMacro(channel, channel.note + macroPtr[1], 0);
+			channel.volume = macroPtr[3];
+			continue;
+
+		case 0x0F:	// Envelope. Parameters: speed, count, endvol
+			channel.envDelta = macroPtr[1];
+			channel.envCount = channel.envSkip = macroPtr[2];
+			channel.envEndVolume = macroPtr[3];
+			continue;
+
+		case 0x11:	// Add Beginn. Parameters: times, Offset(W)
+			channel.addBeginLength = channel.addBeginCount = macroPtr[1];
+			channel.addBeginDelta = (int16)READ_BE_UINT16(&macroPtr[2]);
+			channel.sampleStart += channel.addBeginDelta;
+			Paula::setChannelSampleStart(channel.paulaChannel, getSamplePtr(channel.sampleStart));
+			continue;
+
+		case 0x12:	// Add Length. Parameters: added Length(W)
+			channel.sampleLen += (int16)READ_BE_UINT16(&macroPtr[2]);
+			Paula::setChannelSampleLen(channel.paulaChannel, channel.sampleLen);
+			continue;
+
+		case 0x14:	// Wait key up. Parameters: wait cycles
+			if (channel.keyUp || channel.macroLoopCount == 0) {
+				channel.macroLoopCount = 0xFF;
+				continue;
+			} else if (channel.macroLoopCount == 0xFF)
+				channel.macroLoopCount = macroPtr[3];
+			--channel.macroLoopCount;
+			--channel.macroStep;
+			return;
+
+		case 0x15:	// Subroutine. Parameters: MacroIndex, Macrostep(W)
+			channel.macroReturnOffset = channel.macroOffset;
+			channel.macroReturnStep = channel.macroStep;
+
+			channel.macroOffset = _resource->macroOffset[macroPtr[1] & (kMaxMacroOffsets - 1)];
+			channel.macroStep = READ_BE_UINT16(&macroPtr[2]);
+			// TODO: MI does some weird stuff there. Figure out which varioables need to be set
+			continue;
+
+		case 0x16:	// Return from Sub.
+			channel.macroOffset = channel.macroReturnOffset;
+			channel.macroStep = channel.macroReturnStep;
+			continue;
+
+		case 0x17:	// Set Period. Parameters: Period(W)
+			channel.refPeriod = READ_BE_UINT16(&macroPtr[2]);
+			if (!channel.portaDelta) {
+				channel.period = channel.refPeriod;
+				//Paula::setChannelPeriod(channel.paulaChannel, channel.period);
+			}
+			continue;
+
+		case 0x18: {	// Sampleloop. Parameters: Offset from Samplestart(W)
+			// TODO: MI loads 24 bit, but thats useless?
+			const uint16 temp = /* ((int8)macroPtr[1] << 16) | */ READ_BE_UINT16(&macroPtr[2]);
+			if (macroPtr[1] || (temp & 1))
+				warning("Tfmx: Problematic value for sampleloop: %06X", (macroPtr[1] << 16) | temp);
+			channel.sampleStart += temp & 0xFFFE;
+			channel.sampleLen -= (temp / 2) /* & 0x7FFF */;
+			Paula::setChannelSampleStart(channel.paulaChannel, getSamplePtr(channel.sampleStart));
+			Paula::setChannelSampleLen(channel.paulaChannel, channel.sampleLen);
+			continue;
+		}
+		case 0x19:	// Set One-Shot Sample
+			channel.addBeginLength = 0;
+			channel.sampleStart = 0;
+			channel.sampleLen = 1;
+			Paula::setChannelSampleStart(channel.paulaChannel, getSamplePtr(0));
+			Paula::setChannelSampleLen(channel.paulaChannel, 1);
+			continue;
+
+		case 0x1A:	// Wait on DMA. Parameters: Cycles-1(W) to wait
+			channel.dmaIntCount = READ_BE_UINT16(&macroPtr[2]) + 1;
+			channel.macroRun = false;
+			Paula::setChannelDmaCount(channel.paulaChannel);
+			break;
+
+/*		case 0x1B:	// Random play. Parameters: macro/speed/mode
+			warnMacroUnimplemented(macroPtr, 0);
+			continue;*/
+
+		case 0x1C:	// Branch on Note. Parameters: note/macrostep(W)
+			if (channel.note > macroPtr[1])
+				channel.macroStep = READ_BE_UINT16(&macroPtr[2]);
+			continue;
+
+		case 0x1D:	// Branch on Volume. Parameters: volume/macrostep(W)
+			if (channel.volume > macroPtr[1])
+				channel.macroStep = READ_BE_UINT16(&macroPtr[2]);
+			continue;
+
+/*		case 0x1E:	// Addvol+note. Parameters: note/CONST./volume
+			warnMacroUnimplemented(macroPtr, 0);
+			continue;*/
+
+		case 0x1F:	// AddPrevNote. Parameters: Note, Finetune(W)
+			setNoteMacro(channel, channel.prevNote + macroPtr[1], READ_BE_UINT16(&macroPtr[2]));
+			break;
+
+		case 0x20:	// Signal. Parameters: signalnumber, value(W)
+			if (_playerCtx.numSignals > macroPtr[1])
+				_playerCtx.signal[macroPtr[1]] = READ_BE_UINT16(&macroPtr[2]);
+			continue;
+
+		case 0x21:	// Play macro. Parameters: macro, chan, detune
+			noteCommand(channel.note, macroPtr[1], (channel.relVol << 4) | macroPtr[2], macroPtr[3]);
+			continue;
+
+		// 0x22 - 0x29 are used by Gem`X
+		// 0x30 - 0x34 are used by Carribean Disaster
+
+		default:
+			debug(3, "Tfmx: Macro %02X not supported", macroPtr[0]);
+		}
+		if (!deferWait)
+			return;
+	}
+}
+
+void Tfmx::advancePatterns() {
+startPatterns:
+	int runningPatterns = 0;
+
+	for (int i = 0; i < kNumChannels; ++i) {
+		PatternContext &pattern = _patternCtx[i];
+		const uint8 pattCmd = pattern.command;
+		if (pattCmd < 0x90) {	// execute Patternstep
+			++runningPatterns;
+			if (!pattern.wait) {
+				// issue all Steps for this tick
+				if (patternRun(pattern)) {
+					// we load the next Trackstep Command and then process all Channels again
+					if (trackRun(true))
+						goto startPatterns;
+					else
+						break;
+				}
+
+			} else
+				--pattern.wait;
+
+		} else if (pattCmd == 0xFE) {	// Stop voice in pattern.expose
+			pattern.command = 0xFF;
+			ChannelContext &channel = _channelCtx[pattern.expose & (kNumVoices - 1)];
+			if (!channel.sfxLocked) {
+				haltMacroProgramm(channel);
+				Paula::disableChannel(channel.paulaChannel);
+			}
+		} // else this pattern-Channel is stopped
+	}
+	if (_playerCtx.stopWithLastPattern && !runningPatterns) {
+		stopPaula();
+	}
+}
+
+bool Tfmx::patternRun(PatternContext &pattern) {
+	for (;;) {
+		const byte *const patternPtr = (const byte *)(getPatternPtr(pattern.offset) + pattern.step);
+		++pattern.step;
+		const byte pattCmd = patternPtr[0];
+
+		if (pattCmd < 0xF0) { // Playnote
+			bool doWait = false;
+			byte noteCmd = pattCmd + pattern.expose;
+			byte param3  = patternPtr[3];
+			if (pattCmd < 0xC0) {	// Note
+				if (pattCmd >= 0x80) {	// Wait
+					pattern.wait = param3;
+					param3 = 0;
+					doWait = true;
+				}
+				noteCmd &= 0x3F;
+			}	// else Portamento
+			noteCommand(noteCmd, patternPtr[1], patternPtr[2], param3);
+			if (doWait)
+				return false;
+
+		} else {	// Patterncommand
+			switch (pattCmd & 0xF) {
+			case 0: 	// End Pattern + Next Trackstep
+				pattern.command = 0xFF;
+				--pattern.step;
+				return true;
+
+			case 1: 	// Loop Pattern. Parameters: Loopcount, PatternStep(W)
+				if (pattern.loopCount != 0) {
+					if (pattern.loopCount == 0xFF)
+						pattern.loopCount = patternPtr[1];
+					pattern.step = READ_BE_UINT16(&patternPtr[2]);
+				}
+				--pattern.loopCount;
+				continue;
+
+			case 2: 	// Jump. Parameters: PatternIndex, PatternStep(W)
+				pattern.offset = _resource->patternOffset[patternPtr[1] & (kMaxPatternOffsets - 1)];
+				pattern.step = READ_BE_UINT16(&patternPtr[2]);
+				continue;
+
+			case 3: 	// Wait. Paramters: ticks to wait
+				pattern.wait = patternPtr[1];
+				return false;
+
+			case 14: 	// Stop custompattern
+				// TODO apparently toggles on/off pattern channel 7
+				debug(3, "Tfmx: Encountered 'Stop custompattern' command");
+				// FT
+			case 4: 	// Stop this pattern
+				pattern.command = 0xFF;
+				--pattern.step;
+				// TODO: try figuring out if this was the last Channel?
+				return false;
+
+			case 5: 	// Key Up Signal. Paramters: channel
+				if (!_channelCtx[patternPtr[2] & (kNumVoices - 1)].sfxLocked)
+					_channelCtx[patternPtr[2] & (kNumVoices - 1)].keyUp = true;
+				continue;
+
+			case 6: 	// Vibrato. Parameters: length, channel, rate
+			case 7: 	// Envelope. Parameters: rate, tempo | channel, endVol
+				noteCommand(pattCmd, patternPtr[1], patternPtr[2], patternPtr[3]);
+				continue;
+
+			case 8: 	// Subroutine. Parameters: pattern, patternstep(W)
+				pattern.savedOffset = pattern.offset;
+				pattern.savedStep = pattern.step;
+
+				pattern.offset = _resource->patternOffset[patternPtr[1] & (kMaxPatternOffsets - 1)];
+				pattern.step = READ_BE_UINT16(&patternPtr[2]);
+				continue;
+
+			case 9: 	// Return from Subroutine
+				pattern.offset = pattern.savedOffset;
+				pattern.step = pattern.savedStep;
+				continue;
+
+			case 10:	// fade. Parameters: tempo, endVol
+				initFadeCommand((uint8)patternPtr[1], (int8)patternPtr[3]);
+				continue;
+
+			case 11:	// play pattern. Parameters: patternCmd, channel, expose
+				initPattern(_patternCtx[patternPtr[2] & (kNumChannels - 1)], patternPtr[1], patternPtr[3], _resource->patternOffset[patternPtr[1] & (kMaxPatternOffsets - 1)]);
+				continue;
+
+			case 12: 	// Lock. Parameters: lockFlag, channel, lockTime
+				_channelCtx[patternPtr[2] & (kNumVoices - 1)].sfxLocked = (patternPtr[1] != 0);
+				_channelCtx[patternPtr[2] & (kNumVoices - 1)].sfxLockTime = patternPtr[3];
+				continue;
+
+			case 13: 	// Cue. Parameters: signalnumber, value(W)
+				if (_playerCtx.numSignals > patternPtr[1])
+					_playerCtx.signal[patternPtr[1]] = READ_BE_UINT16(&patternPtr[2]);
+				continue;
+
+			case 15: 	// NOP
+				continue;
+			}
+		}
+	}
+}
+
+bool Tfmx::trackRun(const bool incStep) {
+	assert(_playerCtx.song >= 0);
+	if (incStep) {
+		// TODO Optionally disable looping
+		if (_trackCtx.posInd == _trackCtx.stopInd)
+			_trackCtx.posInd = _trackCtx.startInd;
+		else
+			++_trackCtx.posInd;
+	}
+	for (;;) {
+		const uint16 *const trackData = getTrackPtr(_trackCtx.posInd);
+
+		if (trackData[0] != FROM_BE_16(0xEFFE)) {
+			// 8 commands for Patterns
+			for (int i = 0; i < 8; ++i) {
+				const uint8 *patCmd = (const uint8 *)&trackData[i];
+				// First byte is pattern number
+				const uint8 patNum = patCmd[0];
+				// if highest bit is set then keep previous pattern
+				if (patNum < 0x80) {
+					initPattern(_patternCtx[i], patNum, patCmd[1], _resource->patternOffset[patNum]);
+				} else {
+					_patternCtx[i].command = patNum;
+					_patternCtx[i].expose = (int8)patCmd[1];
+				}
+			}
+			return true;
+
+		} else {
+			// 16 byte Trackstep Command
+			switch (READ_BE_UINT16(&trackData[1])) {
+			case 0:	// Stop Player. No Parameters
+				stopPaula();
+				return false;
+
+			case 1:	// Branch/Loop section of tracksteps. Parameters: branch target, loopcount
+				if (_trackCtx.loopCount != 0) {
+					if (_trackCtx.loopCount < 0)
+						_trackCtx.loopCount = READ_BE_UINT16(&trackData[3]);
+					_trackCtx.posInd    = READ_BE_UINT16(&trackData[2]);
+					continue;
+				}
+				--_trackCtx.loopCount;
+				break;
+
+			case 2:	{ // Set Tempo. Parameters: tempo, divisor
+				_playerCtx.patternCount = _playerCtx.patternSkip = READ_BE_UINT16(&trackData[2]); // tempo
+				const uint16 temp = READ_BE_UINT16(&trackData[3]); // divisor
+
+				if (!(temp & 0x8000) && (temp & 0x1FF))
+					setInterruptFreqUnscaled(temp & 0x1FF);
+				break;
+			}
+			case 4:	// Fade. Parameters: tempo, endVol
+				// load the LSB of the 16bit words
+				initFadeCommand(((const uint8 *)&trackData[2])[1], ((const int8 *)&trackData[3])[1]);
+				break;
+
+			case 3:	// Unknown, stops player aswell
+			default:
+				debug(3, "Tfmx: Unknown Trackstep Command: %02X", READ_BE_UINT16(&trackData[1]));
+				// MI-Player handles this by stopping the player, we just continue
+			}
+		}
+
+		if (_trackCtx.posInd == _trackCtx.stopInd) {
+			warning("Tfmx: Reached invalid Song-Position");
+			return false;
+		}
+		++_trackCtx.posInd;
+	}
+}
+
+void Tfmx::noteCommand(const uint8 note, const uint8 param1, const uint8 param2, const uint8 param3) {
+	ChannelContext &channel = _channelCtx[param2 & (kNumVoices - 1)];
+
+	if (note == 0xFC) {	// Lock command
+		channel.sfxLocked = (param1 != 0);
+		channel.sfxLockTime = param3; // only 1 byte read!
+
+	} else if (channel.sfxLocked) {	// Channel still locked, do nothing
+
+	} else if (note < 0xC0) {	// Play Note - Parameters: note, macro, relVol | channel, finetune
+
+		channel.prevNote = channel.note;
+		channel.note = note;
+		// channel.macroIndex = param1 & (kMaxMacroOffsets - 1);
+		channel.macroOffset = _resource->macroOffset[param1 & (kMaxMacroOffsets - 1)];
+		channel.relVol = param2 >> 4;
+		channel.fineTune = (int8)param3;
+
+		// TODO: the point where the channel gets initialised varies with the games, needs more research.
+		initMacroProgramm(channel);
+		channel.keyUp = false; // key down = playing a Note
+
+	} else if (note < 0xF0) {	// Portamento - Parameters: note, tempo, channel, rate
+		channel.portaSkip = param1;
+		channel.portaCount = 1;
+		if (!channel.portaDelta)
+			channel.portaValue = channel.refPeriod;
+		channel.portaDelta = param3;
+
+		channel.note = note & 0x3F;
+		channel.refPeriod = noteIntervalls[channel.note];
+
+	} else switch (note) {	// Command
+
+		case 0xF5:	// Key Up Signal
+			channel.keyUp = true;
+			break;
+
+		case 0xF6:	// Vibratio - Parameters: length, channel, rate
+			channel.vibLength = param1 & 0xFE;
+			channel.vibCount = param1 / 2;
+			channel.vibDelta = param3;
+			channel.vibValue = 0;
+			break;
+
+		case 0xF7:	// Envelope - Parameters: rate, tempo | channel, endVol
+			channel.envDelta = param1;
+			channel.envCount = channel.envSkip = (param2 >> 4) + 1;
+			channel.envEndVolume = param3;
+			break;
+	}
+}
+
+void Tfmx::initMacroProgramm(ChannelContext &channel) {
+	channel.macroStep = 0;
+	channel.macroWait = 0;
+	channel.macroRun = true;
+	channel.macroSfxRun = 0;
+	channel.macroLoopCount = 0xFF;
+	channel.dmaIntCount = 0;
+	channel.deferWait = false;
+
+	channel.macroReturnOffset = 0;
+	channel.macroReturnStep = 0;
+}
+
+void Tfmx::clearEffects(ChannelContext &channel) {
+	channel.addBeginLength = 0;
+	channel.envSkip = 0;
+	channel.vibLength = 0;
+	channel.portaDelta = 0;
+}
+
+void Tfmx::haltMacroProgramm(ChannelContext &channel) {
+	channel.macroRun = false;
+	channel.dmaIntCount = 0;
+}
+
+void Tfmx::unlockMacroChannel(ChannelContext &channel) {
+	channel.customMacro = 0;
+	channel.customMacroIndex = 0;
+	channel.customMacroPrio = 0;
+	channel.sfxLocked = false;
+	channel.sfxLockTime = -1;
+}
+
+void Tfmx::initPattern(PatternContext &pattern, uint8 cmd, int8 expose, uint32 offset) {
+	pattern.command = cmd;
+	pattern.offset = offset;
+	pattern.expose = expose;
+	pattern.step = 0;
+	pattern.wait = 0;
+	pattern.loopCount = 0xFF;
+
+	pattern.savedOffset = 0;
+	pattern.savedStep = 0;
+}
+
+void Tfmx::stopSongImpl(bool stopAudio) {
+	 _playerCtx.song = -1;
+	for (int i = 0; i < kNumChannels; ++i) {
+		_patternCtx[i].command = 0xFF;
+		_patternCtx[i].expose = 0;
+	}
+	if (stopAudio) {
+		stopPaula();
+		for (int i = 0; i < kNumVoices; ++i) {
+			clearEffects(_channelCtx[i]);
+			unlockMacroChannel(_channelCtx[i]);
+			haltMacroProgramm(_channelCtx[i]);
+			_channelCtx[i].note = 0;
+			_channelCtx[i].volume = 0;
+			_channelCtx[i].macroSfxRun = -1;
+			_channelCtx[i].vibValue = 0;
+
+			_channelCtx[i].sampleStart = 0;
+			_channelCtx[i].sampleLen = 2;
+			_channelCtx[i].refPeriod = 4;
+			_channelCtx[i].period = 4;
+			Paula::disableChannel(i);
+		}
+	}
+}
+
+void Tfmx::setNoteMacro(ChannelContext &channel, uint note, int fineTune) {
+	const uint16 noteInt = noteIntervalls[note & 0x3F];
+	const uint16 finetune = (uint16)(fineTune + channel.fineTune + (1 << 8));
+	channel.refPeriod = ((uint32)noteInt * finetune >> 8);
+	if (!channel.portaDelta)
+		channel.period = channel.refPeriod;
+}
+
+void Tfmx::initFadeCommand(const uint8 fadeTempo, const int8 endVol) {
+	_playerCtx.fadeCount = _playerCtx.fadeSkip = fadeTempo;
+	_playerCtx.fadeEndVolume = endVol;
+
+	if (fadeTempo) {
+		const int diff = _playerCtx.fadeEndVolume - _playerCtx.volume;
+		_playerCtx.fadeDelta = (diff != 0) ? ((diff > 0) ? 1 : -1) : 0;
+	} else {
+		_playerCtx.volume = endVol;
+		_playerCtx.fadeDelta = 0;
+	}
+}
+
+void Tfmx::setModuleData(Tfmx &otherPlayer) {
+	setModuleData(otherPlayer._resource, otherPlayer._resourceSample.sampleData, otherPlayer._resourceSample.sampleLen, false);
+}
+
+bool Tfmx::load(Common::SeekableReadStream &musicData, Common::SeekableReadStream &sampleData, bool autoDelete) {
+	const MdatResource *mdat = loadMdatFile(musicData);
+	if (mdat) {
+		uint32 sampleLen = 0;
+		const int8 *sampleDat = loadSampleFile(sampleLen, sampleData);
+		if (sampleDat) {
+			setModuleData(mdat, sampleDat, sampleLen, autoDelete);
+			return true;
+		}
+		delete[] mdat->mdatAlloc;
+		delete mdat;
+	}
+	return false;
+}
+
+void Tfmx::freeResourceDataImpl() {
+	if (_deleteResource) {
+		if (_resource) {
+			delete[] _resource->mdatAlloc;
+			delete _resource;
+		}
+		delete[] _resourceSample.sampleData;
+	}
+	_resource = 0;
+	_resourceSample.sampleData = 0;
+	_resourceSample.sampleLen = 0;
+	_deleteResource = false;
+}
+
+void Tfmx::setModuleData(const MdatResource *resource, const int8 *sampleData, uint32 sampleLen, bool autoDelete) {
+	Common::StackLock lock(_mutex);
+	stopSongImpl(true);
+	freeResourceDataImpl();
+	_resource = resource;
+	_resourceSample.sampleData = sampleData;
+	_resourceSample.sampleLen = sampleData ? sampleLen : 0;
+	_deleteResource = autoDelete;
+}
+
+const int8 *Tfmx::loadSampleFile(uint32 &sampleLen, Common::SeekableReadStream &sampleStream) {
+	sampleLen = 0;
+
+	const int32 sampleSize = sampleStream.size();
+	if (sampleSize < 4) {
+		warning("Tfmx: Cant load Samplefile");
+		return false;
+	}
+
+	int8 *sampleAlloc = new int8[sampleSize];
+	if (!sampleAlloc) {
+		warning("Tfmx: Could not allocate Memory: %dKB", sampleSize / 1024);
+		return 0;
+	}
+
+	if (sampleStream.read(sampleAlloc, sampleSize) == (uint32)sampleSize) {
+		sampleAlloc[0] = sampleAlloc[1] = sampleAlloc[2] = sampleAlloc[3] = 0;
+		sampleLen = sampleSize;
+	} else {
+		delete[] sampleAlloc;
+		warning("Tfmx: Encountered IO-Error");
+		return 0;
+	}
+	return sampleAlloc;
+}
+
+const Tfmx::MdatResource *Tfmx::loadMdatFile(Common::SeekableReadStream &musicData) {
+	bool hasHeader = false;
+	const int32 mdatSize = musicData.size();
+	if (mdatSize >= 0x200) {
+		byte buf[16] = { 0 };
+		// 0x0000: 10 Bytes Header "TFMX-SONG "
+		musicData.read(buf, 10);
+		hasHeader = memcmp(buf, "TFMX-SONG ", 10) == 0;
+	}
+
+	if (!hasHeader) {
+		warning("Tfmx: File is not a Tfmx Module");
+		return 0;
+	}
+
+	MdatResource *resource = new MdatResource;
+
+	resource->mdatAlloc = 0;
+	resource->mdatData = 0;
+	resource->mdatLen = 0;
+
+	// 0x000A: int16 flags
+	resource->headerFlags = musicData.readUint16BE();
+	// 0x000C: int32 ?
+	// 0x0010: 6*40 Textfield
+	musicData.skip(4 + 6 * 40);
+
+	/* 0x0100: Songstart x 32*/
+	for (int i = 0; i < kNumSubsongs; ++i)
+		resource->subsong[i].songstart = musicData.readUint16BE();
+	/* 0x0140: Songend x 32*/
+	for (int i = 0; i < kNumSubsongs; ++i)
+		resource->subsong[i].songend = musicData.readUint16BE();
+	/* 0x0180: Tempo x 32*/
+	for (int i = 0; i < kNumSubsongs; ++i)
+		resource->subsong[i].tempo  = musicData.readUint16BE();
+
+	/* 0x01c0: unused ? */
+	musicData.skip(16);
+
+	/* 0x01d0: trackstep, pattern data p, macro data p */
+	const uint32 offTrackstep = musicData.readUint32BE();
+	uint32 offPatternP, offMacroP;
+
+	// This is how MI`s TFMX-Player tests for unpacked Modules.
+	if (offTrackstep == 0) { // unpacked File
+		resource->trackstepOffset = 0x600 + 0x200;
+		offPatternP		= 0x200 + 0x200;
+		offMacroP		= 0x400 + 0x200;
+	} else { // packed File
+		resource->trackstepOffset = offTrackstep;
+		offPatternP = musicData.readUint32BE();
+		offMacroP = musicData.readUint32BE();
+	}
+
+	// End of basic header, check if everything worked ok
+	if (musicData.err()) {
+		warning("Tfmx: Encountered IO-Error");
+		delete resource;
+		return 0;
+	}
+
+	// TODO: if a File is packed it could have for Ex only 2 Patterns/Macros
+	// the following loops could then read beyond EOF.
+	// To correctly handle this it would be necessary to sort the pointers and
+	// figure out the number of Macros/Patterns
+	// We could also analyze pointers if they are correct offsets,
+	// so that accesses can be unchecked later
+
+	// Read in pattern starting offsets
+	musicData.seek(offPatternP);
+	for (int i = 0; i < kMaxPatternOffsets; ++i)
+		resource->patternOffset[i] = musicData.readUint32BE();
+
+	// use last PatternOffset (stored at 0x5FC in mdat) if unpacked File
+	// or fixed offset 0x200 if packed
+	resource->sfxTableOffset = offTrackstep ? 0x200 : resource->patternOffset[127];
+
+	// Read in macro starting offsets
+	musicData.seek(offMacroP);
+	for (int i = 0; i < kMaxMacroOffsets; ++i)
+		resource->macroOffset[i] = musicData.readUint32BE();
+
+	// Read in mdat-file
+	// TODO: we can skip everything thats already stored in the resource-structure.
+	const int32 mdatOffset = offTrackstep ? 0x200 : 0x600;	// 0x200 is very conservative
+	const uint32 allocSize = (uint32)mdatSize - mdatOffset;
+
+	byte *mdatAlloc = new byte[allocSize];
+	if (!mdatAlloc) {
+		warning("Tfmx: Could not allocate Memory: %dKB", allocSize / 1024);
+		delete resource;
+		return 0;
+	}
+	musicData.seek(mdatOffset);
+	if (musicData.read(mdatAlloc, allocSize) == allocSize) {
+		resource->mdatAlloc = mdatAlloc;
+		resource->mdatData = mdatAlloc - mdatOffset;
+		resource->mdatLen = mdatSize;
+	} else {
+		delete[] mdatAlloc;
+		warning("Tfmx: Encountered IO-Error");
+		delete resource;
+		return 0;
+	}
+
+	return resource;
+}
+
+void Tfmx::doMacro(int note, int macro, int relVol, int finetune, int channelNo) {
+	assert(0 <= macro && macro < kMaxMacroOffsets);
+	assert(0 <= note && note < 0xC0);
+	Common::StackLock lock(_mutex);
+
+	if (!hasResources())
+		return;
+	channelNo &= (kNumVoices - 1);
+	ChannelContext &channel = _channelCtx[channelNo];
+	unlockMacroChannel(channel);
+
+	noteCommand((uint8)note, (uint8)macro, (uint8)((relVol << 4) | channelNo), (uint8)finetune);
+	startPaula();
+}
+
+void Tfmx::stopMacroEffect(int channel) {
+	assert(0 <= channel && channel < kNumVoices);
+	Common::StackLock lock(_mutex);
+	unlockMacroChannel(_channelCtx[channel]);
+	haltMacroProgramm(_channelCtx[channel]);
+	Paula::disableChannel(_channelCtx[channel].paulaChannel);
+}
+
+void Tfmx::doSong(int songPos, bool stopAudio) {
+	assert(0 <= songPos && songPos < kNumSubsongs);
+	Common::StackLock lock(_mutex);
+
+	stopSongImpl(stopAudio);
+
+	if (!hasResources())
+		return;
+
+	_trackCtx.loopCount = -1;
+	_trackCtx.startInd = _trackCtx.posInd = _resource->subsong[songPos].songstart;
+	_trackCtx.stopInd = _resource->subsong[songPos].songend;
+	_playerCtx.song = (int8)songPos;
+
+	const bool palFlag = (_resource->headerFlags & 2) != 0;
+	const uint16 tempo = _resource->subsong[songPos].tempo;
+	uint16 ciaIntervall;
+	if (tempo >= 0x10) {
+		ciaIntervall = (uint16)(kCiaBaseInterval / tempo);
+		_playerCtx.patternSkip = 0;
+	} else {
+		ciaIntervall = palFlag ? (uint16)kPalDefaultCiaVal : (uint16)kNtscDefaultCiaVal;
+		_playerCtx.patternSkip = tempo;
+	}
+	setInterruptFreqUnscaled(ciaIntervall);
+	Paula::setAudioFilter(true);
+
+	_playerCtx.patternCount = 0;
+	if (trackRun())
+		startPaula();
+}
+
+int Tfmx::doSfx(uint16 sfxIndex, bool unlockChannel) {
+	assert(sfxIndex < 128);
+	Common::StackLock lock(_mutex);
+
+	if (!hasResources())
+		return -1;
+	const byte *sfxEntry = getSfxPtr(sfxIndex);
+	if (sfxEntry[0] == 0xFB) {
+		warning("Tfmx: custom patterns are not supported");
+		// custompattern
+		/* const uint8 patCmd = sfxEntry[2];
+		const int8 patExp = (int8)sfxEntry[3]; */
+	} else {
+		// custommacro
+		const byte channelNo = ((_playerCtx.song >= 0) ? sfxEntry[2] : sfxEntry[4]) & (kNumVoices - 1);
+		const byte priority = sfxEntry[5] & 0x7F;
+
+		ChannelContext &channel = _channelCtx[channelNo];
+		if (unlockChannel)
+			unlockMacroChannel(channel);
+
+		const int16 sfxLocktime = channel.sfxLockTime;
+		if (priority >= channel.customMacroPrio || sfxLocktime < 0) {
+			if (sfxIndex != channel.customMacroIndex || sfxLocktime < 0 || (sfxEntry[5] < 0x80)) {
+				channel.customMacro = READ_UINT32(sfxEntry); // intentionally not "endian-correct"
+				channel.customMacroPrio = priority;
+				channel.customMacroIndex = (uint8)sfxIndex;
+				debug(3, "Tfmx: running Macro %08X on channel %i - priority: %02X", TO_BE_32(channel.customMacro), channelNo, priority);
+				return channelNo;
+			}
+		}
+	}
+	return -1;
+}
+
+}	// End of namespace Audio
+
+// some debugging functions
+#if 0
+namespace {
+
+void displayMacroStep(const void * const vptr) {
+	static const char *tableMacros[] = {
+		"DMAoff+Resetxx/xx/xx flag/addset/vol   ",
+		"DMAon (start sample at selected begin) ",
+		"SetBegin    xxxxxx   sample-startadress",
+		"SetLen      ..xxxx   sample-length     ",
+		"Wait        ..xxxx   count (VBI''s)    ",
+		"Loop        xx/xxxx  count/step        ",
+		"Cont        xx/xxxx  macro-number/step ",
+		"-------------STOP----------------------",
+		"AddNote     xx/xxxx  note/detune       ",
+		"SetNote     xx/xxxx  note/detune       ",
+		"Reset   Vibrato-Portamento-Envelope    ",
+		"Portamento  xx/../xx count/speed       ",
+		"Vibrato     xx/../xx speed/intensity   ",
+		"AddVolume   ....xx   volume 00-3F      ",
+		"SetVolume   ....xx   volume 00-3F      ",
+		"Envelope    xx/xx/xx speed/count/endvol",
+		"Loop key up xx/xxxx  count/step        ",
+		"AddBegin    xx/xxxx  count/add to start",
+		"AddLen      ..xxxx   add to sample-len ",
+		"DMAoff stop sample but no clear        ",
+		"Wait key up ....xx   count (VBI''s)    ",
+		"Go submacro xx/xxxx  macro-number/step ",
+		"--------Return to old macro------------",
+		"Setperiod   ..xxxx   DMA period        ",
+		"Sampleloop  ..xxxx   relative adress   ",
+		"-------Set one shot sample-------------",
+		"Wait on DMA ..xxxx   count (Wavecycles)",
+		"Random play xx/xx/xx macro/speed/mode  ",
+		"Splitkey    xx/xxxx  key/macrostep     ",
+		"Splitvolume xx/xxxx  volume/macrostep  ",
+		"Addvol+note xx/fe/xx note/CONST./volume",
+		"SetPrevNote xx/xxxx  note/detune       ",
+		"Signal      xx/xxxx  signalnumber/value",
+		"Play macro  xx/.x/xx macro/chan/detune ",
+		"SID setbeg  xxxxxx   sample-startadress",
+		"SID setlen  xx/xxxx  buflen/sourcelen  ",
+		"SID op3 ofs xxxxxx   offset            ",
+		"SID op3 frq xx/xxxx  speed/amplitude   ",
+		"SID op2 ofs xxxxxx   offset            ",
+		"SID op2 frq xx/xxxx  speed/amplitude   ",
+		"SID op1     xx/xx/xx speed/amplitude/TC",
+		"SID stop    xx....   flag (1=clear all)"
+	};
+
+	const byte *const macroData = (const byte * const)vptr;
+	if (macroData[0] < ARRAYSIZE(tableMacros))
+		debug("%s %02X%02X%02X", tableMacros[macroData[0]], macroData[1], macroData[2], macroData[3]);
+	else
+		debug("Unknown Macro #%02X %02X%02X%02X", macroData[0], macroData[1], macroData[2], macroData[3]);
+}
+
+void displayPatternstep(const void * const vptr) {
+	static const char *tablePatterns[] = {
+		"End --Next track  step--",
+		"Loop[count     / step.w]",
+		"Cont[patternno./ step.w]",
+		"Wait[count 00-FF--------",
+		"Stop--Stop this pattern-",
+		"Kup^-Set key up/channel]",
+		"Vibr[speed     / rate.b]",
+		"Enve[speed /endvolume.b]",
+		"GsPt[patternno./ step.w]",
+		"RoPt-Return old pattern-",
+		"Fade[speed /endvolume.b]",
+		"PPat[patt./track+transp]",
+		"Lock---------ch./time.b]",
+		"Cue [number.b/  value.w]",
+		"Stop-Stop custompattern-",
+		"NOP!-no operation-------"
+	};
+
+	const byte * const patData = (const byte * const)vptr;
+	const byte command = patData[0];
+	if (command < 0xF0) { // Playnote
+		const byte flags = command >> 6; // 0-1 means note+detune, 2 means wait, 3 means portamento?
+		const char *flagsSt[] = { "Note ", "Note ", "Wait ", "Porta" };
+		debug("%s %02X%02X%02X%02X", flagsSt[flags], patData[0], patData[1], patData[2], patData[3]);
+	} else
+		debug("%s %02X%02X%02X",tablePatterns[command & 0xF], patData[1], patData[2], patData[3]);
+}
+
+} // End of anonymous namespace
+#endif
+
+#endif // #if defined(SOUND_MODS_TFMX_H)
diff --git a/audio/mods/tfmx.h b/audio/mods/tfmx.h
new file mode 100644
index 0000000000..1930487eb8
--- /dev/null
+++ b/audio/mods/tfmx.h
@@ -0,0 +1,284 @@
+/* ScummVM - Graphic Adventure Engine
+ *
+ * ScummVM is the legal property of its developers, whose names
+ * are too numerous to list here. Please refer to the COPYRIGHT
+ * file distributed with this source distribution.
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
+ *
+ * $URL$
+ * $Id$
+ *
+ */
+
+// see if all engines using this class are DISABLED
+#if !defined(ENABLE_SCUMM)
+
+// normal Header Guard
+#elif !defined(SOUND_MODS_TFMX_H)
+#define SOUND_MODS_TFMX_H
+
+#include "audio/mods/paula.h"
+
+namespace Audio {
+
+class Tfmx : public Paula {
+public:
+	Tfmx(int rate, bool stereo);
+	virtual ~Tfmx();
+
+	/**
+	 * Stops a playing Song (but leaves macros running) and optionally also stops the player
+	 *
+	 * @param stopAudio	stops player and audio output
+	 * @param dataSize	number of bytes to be written
+	 * @return the number of bytes which were actually written.
+	 */
+	void stopSong(bool stopAudio = true) { Common::StackLock lock(_mutex); stopSongImpl(stopAudio); }
+	/**
+	 * Stops currently playing Song (if any) and cues up a new one.
+	 * if stopAudio is specified, the player gets reset before starting the new song
+	 *
+	 * @param songPos	index of Song to play
+	 * @param stopAudio	stops player and audio output
+	 * @param dataSize	number of bytes to be written
+	 * @return the number of bytes which were actually written.
+	 */
+	void doSong(int songPos, bool stopAudio = false);
+	/**
+	 * plays an effect from the sfx-table, does not start audio-playback.
+	 *
+	 * @param sfxIndex	index of effect to play
+	 * @param unlockChannel	overwrite higher priority effects
+	 * @return	index of the channel which now queued up the effect.
+	 *			-1 in case the effect couldnt be queued up
+	 */
+	int doSfx(uint16 sfxIndex, bool unlockChannel = false);
+	/**
+	 * stop a running macro channel
+	 *
+	 * @param channel	index of effect to stop
+	 */
+	void stopMacroEffect(int channel);
+
+	void doMacro(int note, int macro, int relVol = 0, int finetune = 0, int channelNo = 0);
+	int getTicks() const { return _playerCtx.tickCount; }
+	int getSongIndex() const { return _playerCtx.song; }
+	void setSignalPtr(uint16 *ptr, uint16 numSignals) { _playerCtx.signal = ptr; _playerCtx.numSignals = numSignals; }
+	void freeResources() { _deleteResource = true; freeResourceDataImpl(); }
+	bool load(Common::SeekableReadStream &musicData, Common::SeekableReadStream &sampleData, bool autoDelete = true);
+	void setModuleData(Tfmx &otherPlayer);
+
+protected:
+	void interrupt();
+
+private:
+	enum { kPalDefaultCiaVal = 11822, kNtscDefaultCiaVal = 14320, kCiaBaseInterval = 0x1B51F8 };
+	enum { kNumVoices = 4, kNumChannels = 8, kNumSubsongs = 32, kMaxPatternOffsets = 128, kMaxMacroOffsets = 128 };
+
+	struct MdatResource {
+		const byte *mdatAlloc;	///< allocated Block of Memory
+		const byte *mdatData;  	///< Start of mdat-File, might point before mdatAlloc to correct Offset
+		uint32 mdatLen;
+
+		uint16 headerFlags;
+//		uint32 headerUnknown;
+//		char textField[6 * 40];
+
+		struct Subsong {
+			uint16 songstart;	///< Index in Trackstep-Table
+			uint16 songend;		///< Last index in Trackstep-Table
+			uint16 tempo;
+		} subsong[kNumSubsongs];
+
+		uint32 trackstepOffset;	///< Offset in mdat
+		uint32 sfxTableOffset;
+
+		uint32 patternOffset[kMaxPatternOffsets];	///< Offset in mdat
+		uint32 macroOffset[kMaxMacroOffsets];	///< Offset in mdat
+
+		void boundaryCheck(const void *address, size_t accessLen = 1) const {
+			assert(mdatAlloc <= address && (const byte *)address + accessLen <= (const byte *)mdatData + mdatLen);
+		}
+	} const *_resource;
+
+	struct SampleResource {
+		const int8 *sampleData;	///< The whole sample-File
+		uint32 sampleLen;
+
+		void boundaryCheck(const void *address, size_t accessLen = 2) const {
+			assert(sampleData <= address && (const byte *)address + accessLen <= (const byte *)sampleData + sampleLen);
+		}
+	} _resourceSample;
+
+	bool _deleteResource;
+
+	bool hasResources() {
+		return _resource && _resource->mdatLen && _resourceSample.sampleLen;
+	}
+
+	struct ChannelContext {
+		byte	paulaChannel;
+
+//		byte	macroIndex;
+		uint16	macroWait;
+		uint32	macroOffset;
+		uint32	macroReturnOffset;
+		uint16	macroStep;
+		uint16	macroReturnStep;
+		uint8	macroLoopCount;
+		bool	macroRun;
+		int8	macroSfxRun;	///< values are the folowing: -1 macro disabled, 0 macro init, 1 macro running
+
+		uint32	customMacro;
+		uint8	customMacroIndex;
+		uint8	customMacroPrio;
+
+		bool	sfxLocked;
+		int16	sfxLockTime;
+		bool	keyUp;
+
+		bool	deferWait;
+		uint16	dmaIntCount;
+
+		uint32	sampleStart;
+		uint16	sampleLen;
+		uint16	refPeriod;
+		uint16	period;
+
+		int8	volume;
+		uint8	relVol;
+		uint8	note;
+		uint8	prevNote;
+		int16	fineTune; // always a signextended byte
+
+		uint8	portaSkip;
+		uint8	portaCount;
+		uint16	portaDelta;
+		uint16	portaValue;
+
+		uint8	envSkip;
+		uint8	envCount;
+		uint8	envDelta;
+		int8	envEndVolume;
+
+		uint8	vibLength;
+		uint8	vibCount;
+		int16	vibValue;
+		int8	vibDelta;
+
+		uint8	addBeginLength;
+		uint8	addBeginCount;
+		int32	addBeginDelta;
+	} _channelCtx[kNumVoices];
+
+	struct PatternContext {
+		uint32	offset; // patternStart, Offset from mdat
+		uint32	savedOffset;	// for subroutine calls
+		uint16	step;	// distance from patternStart
+		uint16	savedStep;
+
+		uint8	command;
+		int8	expose;
+		uint8	loopCount;
+		uint8	wait;	///< how many ticks to wait before next Command
+	} _patternCtx[kNumChannels];
+
+	struct TrackStepContext {
+		uint16	startInd;
+		uint16	stopInd;
+		uint16	posInd;
+		int16	loopCount;
+	} _trackCtx;
+
+	struct PlayerContext {
+		int8	song;	///< >= 0 if Song is running (means process Patterns)
+
+		uint16	patternCount;
+		uint16	patternSkip;	///< skip that amount of CIA-Interrupts
+
+		int8	volume;	///< Master Volume
+
+		uint8	fadeSkip;
+		uint8	fadeCount;
+		int8	fadeEndVolume;
+		int8	fadeDelta;
+
+		int		tickCount;
+
+		uint16	*signal;
+		uint16	numSignals;
+
+		bool	stopWithLastPattern; ///< hack to automatically stop the whole player if no Pattern is running
+	} _playerCtx;
+
+	const byte *getSfxPtr(uint16 index = 0) const {
+		const byte *sfxPtr = (const byte *)(_resource->mdatData + _resource->sfxTableOffset + index * 8);
+
+		_resource->boundaryCheck(sfxPtr, 8);
+		return sfxPtr;
+	}
+
+	const uint16 *getTrackPtr(uint16 trackstep = 0) const {
+		const uint16 *trackData = (const uint16 *)(_resource->mdatData + _resource->trackstepOffset + 16 * trackstep);
+
+		_resource->boundaryCheck(trackData, 16);
+		return trackData;
+	}
+
+	const uint32 *getPatternPtr(uint32 offset) const {
+		const uint32 *pattData = (const uint32 *)(_resource->mdatData + offset);
+
+		_resource->boundaryCheck(pattData, 4);
+		return pattData;
+	}
+
+	const uint32 *getMacroPtr(uint32 offset) const {
+		const uint32 *macroData = (const uint32 *)(_resource->mdatData + offset);
+
+		_resource->boundaryCheck(macroData, 4);
+		return macroData;
+	}
+
+	const int8 *getSamplePtr(const uint32 offset) const {
+		const int8 *sample = _resourceSample.sampleData + offset;
+
+		_resourceSample.boundaryCheck(sample, 2);
+		return sample;
+	}
+
+	static inline void initMacroProgramm(ChannelContext &channel);
+	static inline void clearEffects(ChannelContext &channel);
+	static inline void haltMacroProgramm(ChannelContext &channel);
+	static inline void unlockMacroChannel(ChannelContext &channel);
+	static inline void initPattern(PatternContext &pattern, uint8 cmd, int8 expose, uint32 offset);
+	void stopSongImpl(bool stopAudio = true);
+	static inline void setNoteMacro(ChannelContext &channel, uint note, int fineTune);
+	void initFadeCommand(const uint8 fadeTempo, const int8 endVol);
+	void setModuleData(const MdatResource *resource, const int8 *sampleData, uint32 sampleLen, bool autoDelete = true);
+	static const MdatResource *loadMdatFile(Common::SeekableReadStream &musicData);
+	static const int8 *loadSampleFile(uint32 &sampleLen, Common::SeekableReadStream &sampleStream);
+	void freeResourceDataImpl();
+	void effects(ChannelContext &channel);
+	void macroRun(ChannelContext &channel);
+	void advancePatterns();
+	bool patternRun(PatternContext &pattern);
+	bool trackRun(bool incStep = false);
+	void noteCommand(uint8 note, uint8 param1, uint8 param2, uint8 param3);
+};
+
+}	// End of namespace Audio
+
+#endif // !defined(SOUND_MODS_TFMX_H)