AUDIO: Rename sound/ dir to audio/

svn-id: r55850

111 files changed, 0 insertions, 41887 deletions

diff --git a/sound/audiostream.cpp b/sound/audiostream.cpp
deleted file mode 100644
index b3efb2cefe..0000000000
--- a/sound/audiostream.cpp
+++ /dev/null
@@ -1,407 +0,0 @@
-/* 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 "common/file.h"
-#include "common/queue.h"
-#include "common/util.h"
-
-#include "sound/audiostream.h"
-#include "sound/decoders/flac.h"
-#include "sound/mixer.h"
-#include "sound/decoders/mp3.h"
-#include "sound/decoders/raw.h"
-#include "sound/decoders/vorbis.h"
-
-
-namespace Audio {
-
-struct StreamFileFormat {
-	/** Decodername */
-	const char *decoderName;
-	const char *fileExtension;
-	/**
-	 * Pointer to a function which tries to open a file of type StreamFormat.
-	 * Return NULL in case of an error (invalid/nonexisting file).
-	 */
-	SeekableAudioStream *(*openStreamFile)(Common::SeekableReadStream *stream, DisposeAfterUse::Flag disposeAfterUse);
-};
-
-static const StreamFileFormat STREAM_FILEFORMATS[] = {
-	/* decoderName,  fileExt, openStreamFuntion */
-#ifdef USE_FLAC
-	{ "FLAC",         ".flac", makeFLACStream },
-	{ "FLAC",         ".fla",  makeFLACStream },
-#endif
-#ifdef USE_VORBIS
-	{ "Ogg Vorbis",   ".ogg",  makeVorbisStream },
-#endif
-#ifdef USE_MAD
-	{ "MPEG Layer 3", ".mp3",  makeMP3Stream },
-#endif
-
-	{ NULL, NULL, NULL } // Terminator
-};
-
-SeekableAudioStream *SeekableAudioStream::openStreamFile(const Common::String &basename) {
-	SeekableAudioStream *stream = NULL;
-	Common::File *fileHandle = new Common::File();
-
-	for (int i = 0; i < ARRAYSIZE(STREAM_FILEFORMATS)-1 && stream == NULL; ++i) {
-		Common::String filename = basename + STREAM_FILEFORMATS[i].fileExtension;
-		fileHandle->open(filename);
-		if (fileHandle->isOpen()) {
-			// Create the stream object
-			stream = STREAM_FILEFORMATS[i].openStreamFile(fileHandle, DisposeAfterUse::YES);
-			fileHandle = 0;
-			break;
-		}
-	}
-
-	delete fileHandle;
-
-	if (stream == NULL)
-		debug(1, "SeekableAudioStream::openStreamFile: Could not open compressed AudioFile %s", basename.c_str());
-
-	return stream;
-}
-
-#pragma mark -
-#pragma mark --- LoopingAudioStream ---
-#pragma mark -
-
-LoopingAudioStream::LoopingAudioStream(RewindableAudioStream *stream, uint loops, DisposeAfterUse::Flag disposeAfterUse)
-    : _parent(stream), _disposeAfterUse(disposeAfterUse), _loops(loops), _completeIterations(0) {
-	assert(stream);
-
-	if (!stream->rewind()) {
-		// TODO: Properly indicate error
-		_loops = _completeIterations = 1;
-	}
-}
-
-LoopingAudioStream::~LoopingAudioStream() {
-	if (_disposeAfterUse == DisposeAfterUse::YES)
-		delete _parent;
-}
-
-int LoopingAudioStream::readBuffer(int16 *buffer, const int numSamples) {
-	if ((_loops && _completeIterations == _loops) || !numSamples)
-		return 0;
-
-	int samplesRead = _parent->readBuffer(buffer, numSamples);
-
-	if (_parent->endOfStream()) {
-		++_completeIterations;
-		if (_completeIterations == _loops)
-			return samplesRead;
-
-		const int remainingSamples = numSamples - samplesRead;
-
-		if (!_parent->rewind()) {
-			// TODO: Properly indicate error
-			_loops = _completeIterations = 1;
-			return samplesRead;
-		}
-
-		return samplesRead + readBuffer(buffer + samplesRead, remainingSamples);
-	}
-
-	return samplesRead;
-}
-
-bool LoopingAudioStream::endOfData() const {
-	return (_loops != 0 && (_completeIterations == _loops));
-}
-
-AudioStream *makeLoopingAudioStream(RewindableAudioStream *stream, uint loops) {
-	if (loops != 1)
-		return new LoopingAudioStream(stream, loops);
-	else
-		return stream;
-}
-
-AudioStream *makeLoopingAudioStream(SeekableAudioStream *stream, Timestamp start, Timestamp end, uint loops) {
-	if (!start.totalNumberOfFrames() && (!end.totalNumberOfFrames() || end == stream->getLength())) {
-		return makeLoopingAudioStream(stream, loops);
-	} else {
-		if (!end.totalNumberOfFrames())
-			end = stream->getLength();
-
-		if (start >= end) {
-			warning("makeLoopingAudioStream: start (%d) >= end (%d)", start.msecs(), end.msecs());
-			delete stream;
-			return 0;
-		}
-
-		return makeLoopingAudioStream(new SubSeekableAudioStream(stream, start, end), loops);
-	}
-}
-
-#pragma mark -
-#pragma mark --- SubLoopingAudioStream ---
-#pragma mark -
-
-SubLoopingAudioStream::SubLoopingAudioStream(SeekableAudioStream *stream,
-                                             uint loops,
-                                             const Timestamp loopStart,
-                                             const Timestamp loopEnd,
-                                             DisposeAfterUse::Flag disposeAfterUse)
-    : _parent(stream), _disposeAfterUse(disposeAfterUse), _loops(loops),
-      _pos(0, getRate() * (isStereo() ? 2 : 1)),
-      _loopStart(convertTimeToStreamPos(loopStart, getRate(), isStereo())),
-      _loopEnd(convertTimeToStreamPos(loopEnd, getRate(), isStereo())),
-      _done(false) {
-	assert(loopStart < loopEnd);
-
-	if (!_parent->rewind())
-		_done = true;
-}
-
-SubLoopingAudioStream::~SubLoopingAudioStream() {
-	if (_disposeAfterUse == DisposeAfterUse::YES)
-		delete _parent;
-}
-
-int SubLoopingAudioStream::readBuffer(int16 *buffer, const int numSamples) {
-	if (_done)
-		return 0;
-
-	int framesLeft = MIN(_loopEnd.frameDiff(_pos), numSamples);
-	int framesRead = _parent->readBuffer(buffer, framesLeft);
-	_pos = _pos.addFrames(framesRead);
-
-	if (framesRead < framesLeft && _parent->endOfData()) {
-		// TODO: Proper error indication.
-		_done = true;
-		return framesRead;
-	} else if (_pos == _loopEnd) {
-		if (_loops != 0) {
-			--_loops;
-			if (!_loops) {
-				_done = true;
-				return framesRead;
-			}
-		}
-
-		if (!_parent->seek(_loopStart)) {
-			// TODO: Proper error indication.
-			_done = true;
-			return framesRead;
-		}
-
-		_pos = _loopStart;
-		framesLeft = numSamples - framesLeft;
-		return framesRead + readBuffer(buffer + framesRead, framesLeft);
-	} else {
-		return framesRead;
-	}
-}
-
-#pragma mark -
-#pragma mark --- SubSeekableAudioStream ---
-#pragma mark -
-
-SubSeekableAudioStream::SubSeekableAudioStream(SeekableAudioStream *parent, const Timestamp start, const Timestamp end, DisposeAfterUse::Flag disposeAfterUse)
-    : _parent(parent), _disposeAfterUse(disposeAfterUse),
-      _start(convertTimeToStreamPos(start, getRate(), isStereo())),
-       _pos(0, getRate() * (isStereo() ? 2 : 1)),
-      _length(convertTimeToStreamPos(end - start, getRate(), isStereo())) {
-
-	assert(_length.totalNumberOfFrames() % (isStereo() ? 2 : 1) == 0);
-	_parent->seek(_start);
-}
-
-SubSeekableAudioStream::~SubSeekableAudioStream() {
-	if (_disposeAfterUse)
-		delete _parent;
-}
-
-int SubSeekableAudioStream::readBuffer(int16 *buffer, const int numSamples) {
-	int framesLeft = MIN(_length.frameDiff(_pos), numSamples);
-	int framesRead = _parent->readBuffer(buffer, framesLeft);
-	_pos = _pos.addFrames(framesRead);
-	return framesRead;
-}
-
-bool SubSeekableAudioStream::seek(const Timestamp &where) {
-	_pos = convertTimeToStreamPos(where, getRate(), isStereo());
-	if (_pos > _length) {
-		_pos = _length;
-		return false;
-	}
-
-	if (_parent->seek(_pos + _start)) {
-		return true;
-	} else {
-		_pos = _length;
-		return false;
-	}
-}
-
-#pragma mark -
-#pragma mark --- Queueing audio stream ---
-#pragma mark -
-
-
-void QueuingAudioStream::queueBuffer(byte *data, uint32 size, DisposeAfterUse::Flag disposeAfterUse, byte flags) {
-	AudioStream *stream = makeRawStream(data, size, getRate(), flags, disposeAfterUse);
-	queueAudioStream(stream, DisposeAfterUse::YES);
-}
-
-
-class QueuingAudioStreamImpl : public QueuingAudioStream {
-private:
-	/**
-	 * We queue a number of (pointers to) audio stream objects.
-	 * In addition, we need to remember for each stream whether
-	 * to dispose it after all data has been read from it.
-	 * Hence, we don't store pointers to stream objects directly,
-	 * but rather StreamHolder structs.
-	 */
-	struct StreamHolder {
-		AudioStream *_stream;
-		DisposeAfterUse::Flag _disposeAfterUse;
-		StreamHolder(AudioStream *stream, DisposeAfterUse::Flag disposeAfterUse)
-		    : _stream(stream),
-		      _disposeAfterUse(disposeAfterUse) {}
-	};
-
-	/**
-	 * The sampling rate of this audio stream.
-	 */
-	const int _rate;
-
-	/**
-	 * Whether this audio stream is mono (=false) or stereo (=true).
-	 */
-	const int _stereo;
-
-	/**
-	 * This flag is set by the finish() method only. See there for more details.
-	 */
-	bool _finished;
-
-	/**
-	 * A mutex to avoid access problems (causing e.g. corruption of
-	 * the linked list) in thread aware environments.
-	 */
-	Common::Mutex _mutex;
-
-	/**
-	 * The queue of audio streams.
-	 */
-	Common::Queue<StreamHolder> _queue;
-
-public:
-	QueuingAudioStreamImpl(int rate, bool stereo)
-	    : _rate(rate), _stereo(stereo), _finished(false) {}
-	~QueuingAudioStreamImpl();
-
-	// Implement the AudioStream API
-	virtual int readBuffer(int16 *buffer, const int numSamples);
-	virtual bool isStereo() const { return _stereo; }
-	virtual int getRate() const { return _rate; }
-	virtual bool endOfData() const {
-		//Common::StackLock lock(_mutex);
-		return _queue.empty();
-	}
-	virtual bool endOfStream() const { return _finished && _queue.empty(); }
-
-	// Implement the QueuingAudioStream API
-	virtual void queueAudioStream(AudioStream *stream, DisposeAfterUse::Flag disposeAfterUse);
-	virtual void finish() { _finished = true; }
-
-	uint32 numQueuedStreams() const {
-		//Common::StackLock lock(_mutex);
-		return _queue.size();
-	}
-};
-
-QueuingAudioStreamImpl::~QueuingAudioStreamImpl() {
-	while (!_queue.empty()) {
-		StreamHolder tmp = _queue.pop();
-		if (tmp._disposeAfterUse == DisposeAfterUse::YES)
-			delete tmp._stream;
-	}
-}
-
-void QueuingAudioStreamImpl::queueAudioStream(AudioStream *stream, DisposeAfterUse::Flag disposeAfterUse) {
-	assert(!_finished);
-	if ((stream->getRate() != getRate()) || (stream->isStereo() != isStereo()))
-		error("QueuingAudioStreamImpl::queueAudioStream: stream has mismatched parameters");
-
-	Common::StackLock lock(_mutex);
-	_queue.push(StreamHolder(stream, disposeAfterUse));
-}
-
-int QueuingAudioStreamImpl::readBuffer(int16 *buffer, const int numSamples) {
-	Common::StackLock lock(_mutex);
-	int samplesDecoded = 0;
-
-	while (samplesDecoded < numSamples && !_queue.empty()) {
-		AudioStream *stream = _queue.front()._stream;
-		samplesDecoded += stream->readBuffer(buffer + samplesDecoded, numSamples - samplesDecoded);
-
-		if (stream->endOfData()) {
-			StreamHolder tmp = _queue.pop();
-			if (tmp._disposeAfterUse == DisposeAfterUse::YES)
-				delete stream;
-		}
-	}
-
-	return samplesDecoded;
-}
-
-QueuingAudioStream *makeQueuingAudioStream(int rate, bool stereo) {
-	return new QueuingAudioStreamImpl(rate, stereo);
-}
-
-Timestamp convertTimeToStreamPos(const Timestamp &where, int rate, bool isStereo) {
-	Timestamp result(where.convertToFramerate(rate * (isStereo ? 2 : 1)));
-
-	// When the Stream is a stereo stream, we have to assure
-	// that the sample position is an even number.
-	if (isStereo && (result.totalNumberOfFrames() & 1))
-		result = result.addFrames(-1); // We cut off one sample here.
-
-	// Since Timestamp allows sub-frame-precision it might lead to odd behaviors
-	// when we would just return result.
-	//
-	// An example is when converting the timestamp 500ms to a 11025 Hz based
-	// stream. It would have an internal frame counter of 5512.5. Now when
-	// doing calculations at frame precision, this might lead to unexpected
-	// results: The frame difference between a timestamp 1000ms and the above
-	// mentioned timestamp (both with 11025 as framerate) would be 5512,
-	// instead of 5513, which is what a frame-precision based code would expect.
-	//
-	// By creating a new Timestamp with the given parameters, we create a
-	// Timestamp with frame-precision, which just drops a sub-frame-precision
-	// information (i.e. rounds down).
-	return Timestamp(result.secs(), result.numberOfFrames(), result.framerate());
-}
-
-} // End of namespace Audio
diff --git a/sound/audiostream.h b/sound/audiostream.h
deleted file mode 100644
index b2c012841d..0000000000
--- a/sound/audiostream.h
+++ /dev/null
@@ -1,371 +0,0 @@
-/* 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_AUDIOSTREAM_H
-#define SOUND_AUDIOSTREAM_H
-
-#include "common/util.h"
-#include "common/scummsys.h"
-#include "common/types.h"
-
-#include "sound/timestamp.h"
-
-namespace Audio {
-
-class SeekableAudioStream;
-
-/**
- * Generic audio input stream. Subclasses of this are used to feed arbitrary
- * sampled audio data into ScummVM's audio mixer.
- */
-class AudioStream {
-public:
-	virtual ~AudioStream() {}
-
-	/**
-	 * Fill the given buffer with up to numSamples samples. Returns the actual
-	 * number of samples read, or -1 if a critical error occurred (note: you
-	 * *must* check if this value is less than what you requested, this can
-	 * happen when the stream is fully used up).
-	 *
-	 * Data has to be in native endianess, 16 bit per sample, signed. For stereo
-	 * stream, buffer will be filled with interleaved left and right channel
-	 * samples, starting with a left sample. Furthermore, the samples in the
-	 * left and right are summed up. So if you request 4 samples from a stereo
-	 * stream, you will get a total of two left channel and two right channel
-	 * samples.
-	 */
-	virtual int readBuffer(int16 *buffer, const int numSamples) = 0;
-
-	/** Is this a stereo stream? */
-	virtual bool isStereo() const = 0;
-
-	/** Sample rate of the stream. */
-	virtual int getRate() const = 0;
-
-	/**
-	 * End of data reached? If this returns true, it means that at this
-	 * time there is no data available in the stream. However there may be
-	 * more data in the future.
-	 * This is used by e.g. a rate converter to decide whether to keep on
-	 * converting data or stop.
-	 */
-	virtual bool endOfData() const = 0;
-
-	/**
-	 * End of stream reached? If this returns true, it means that all data
-	 * in this stream is used up and no additional data will appear in it
-	 * in the future.
-	 * This is used by the mixer to decide whether a given stream shall be
-	 * removed from the list of active streams (and thus be destroyed).
-	 * By default this maps to endOfData()
-	 */
-	virtual bool endOfStream() const { return endOfData(); }
-};
-
-/**
- * A rewindable audio stream. This allows for reseting the AudioStream
- * to its initial state. Note that rewinding itself is not required to
- * be working when the stream is being played by Mixer!
- */
-class RewindableAudioStream : public AudioStream {
-public:
-	/**
-	 * Rewinds the stream to its start.
-	 *
-	 * @return true on success, false otherwise.
-	 */
-	virtual bool rewind() = 0;
-};
-
-/**
- * A looping audio stream. This object does nothing besides using
- * a RewindableAudioStream to play a stream in a loop.
- */
-class LoopingAudioStream : public AudioStream {
-public:
-	/**
-	 * Creates a looping audio stream object.
-	 *
-	 * Note that on creation of the LoopingAudioStream object
-	 * the underlying stream will be rewound.
-	 *
-	 * @see makeLoopingAudioStream
-	 *
-	 * @param stream Stream to loop
-	 * @param loops How often to loop (0 = infinite)
-	 * @param disposeAfterUse Destroy the stream after the LoopingAudioStream has finished playback.
-	 */
-	LoopingAudioStream(RewindableAudioStream *stream, uint loops, DisposeAfterUse::Flag disposeAfterUse = DisposeAfterUse::YES);
-	~LoopingAudioStream();
-
-	int readBuffer(int16 *buffer, const int numSamples);
-	bool endOfData() const;
-
-	bool isStereo() const { return _parent->isStereo(); }
-	int getRate() const { return _parent->getRate(); }
-
-	/**
-	 * Returns number of loops the stream has played.
-	 *
-	 * @param numLoops number of loops to play, 0 - infinite
-	 */
-	uint getCompleteIterations() const { return _completeIterations; }
-private:
-	RewindableAudioStream *_parent;
-	DisposeAfterUse::Flag _disposeAfterUse;
-
-	uint _loops;
-	uint _completeIterations;
-};
-
-/**
- * Wrapper functionality to efficiently create a stream, which might be looped.
- *
- * Note that this function does not return a LoopingAudioStream, because it does
- * not create one when the loop count is "1". This allows to keep the runtime
- * overhead down, when the code does not require any functionality only offered
- * by LoopingAudioStream.
- *
- * @param stream  Stream to loop (will be automatically destroyed, when the looping is done)
- * @param loops   How often to loop (0 = infinite)
- * @return A new AudioStream, which offers the desired functionality.
- */
-AudioStream *makeLoopingAudioStream(RewindableAudioStream *stream, uint loops);
-
-/**
- * A seekable audio stream. Subclasses of this class implement an
- * interface for seeking. The seeking itself is not required to be
- * working while the stream is being played by Mixer!
- */
-class SeekableAudioStream : public RewindableAudioStream {
-public:
-	/**
-	 * Tries to load a file by trying all available formats.
-	 * In case of an error, the file handle will be closed, but deleting
-	 * it is still the responsibility of the caller.
-	 *
-	 * @param basename a filename without an extension
-	 * @return  an SeekableAudioStream ready to use in case of success;
-	 *          NULL in case of an error (e.g. invalid/nonexisting file)
-	 */
-	static SeekableAudioStream *openStreamFile(const Common::String &basename);
-
-	/**
-	 * Seeks to a given offset in the stream.
-	 *
-	 * @param where offset in milliseconds
-	 * @return true on success, false on failure.
-	 */
-	bool seek(uint32 where) {
-		return seek(Timestamp(where, getRate()));
-	}
-
-	/**
-	 * Seeks to a given offset in the stream.
-	 *
-	 * @param where offset as timestamp
-	 * @return true on success, false on failure.
-	 */
-	virtual bool seek(const Timestamp &where) = 0;
-
-	/**
-	 * Returns the length of the stream.
-	 *
-	 * @return length as Timestamp.
-	 */
-	virtual Timestamp getLength() const = 0;
-
-	virtual bool rewind() { return seek(0); }
-};
-
-/**
- * Wrapper functionality to efficiently create a stream, which might be looped
- * in a certain interval.
- *
- * This automatically starts the stream at time "start"!
- *
- * Note that this function does not return a LoopingAudioStream, because it does
- * not create one when the loop count is "1". This allows to keep the runtime
- * overhead down, when the code does not require any functionality only offered
- * by LoopingAudioStream.
- *
- * @param stream Stream to loop (will be automatically destroyed, when the looping is done)
- * @param start  Starttime of the stream interval to be looped
- * @param end    End of the stream interval to be looped (a zero time, means till end)
- * @param loops  How often to loop (0 = infinite)
- * @return A new AudioStream, which offers the desired functionality.
- */
-AudioStream *makeLoopingAudioStream(SeekableAudioStream *stream, Timestamp start, Timestamp end, uint loops);
-
-/**
- * A looping audio stream, which features looping of a nested part of the
- * stream.
- *
- * NOTE:
- * Currently this implementation stops after the nested loop finished
- * playback.
- *
- * IMPORTANT:
- * This might be merged with SubSeekableAudioStream for playback purposes.
- * (After extending it to accept a start time).
- */
-class SubLoopingAudioStream : public AudioStream {
-public:
-	/**
-	 * Constructor for a SubLoopingAudioStream.
-	 *
-	 * Note that on creation of the SubLoopingAudioStream object
-	 * the underlying stream will be rewound.
-	 *
-	 * @param stream          Stream to loop
-	 * @param loops           How often the stream should be looped (0 means infinite)
-	 * @param loopStart       Start of the loop (this must be smaller than loopEnd)
-	 * @param loopEnd         End of the loop (thus must be greater than loopStart)
-	 * @param disposeAfterUse Whether the stream should be disposed, when the
-	 *                        SubLoopingAudioStream is destroyed.
-	 */
-	SubLoopingAudioStream(SeekableAudioStream *stream, uint loops,
-	                      const Timestamp loopStart,
-	                      const Timestamp loopEnd,
-	                      DisposeAfterUse::Flag disposeAfterUse = DisposeAfterUse::YES);
-	~SubLoopingAudioStream();
-
-	int readBuffer(int16 *buffer, const int numSamples);
-	bool endOfData() const { return _done; }
-
-	bool isStereo() const { return _parent->isStereo(); }
-	int getRate() const { return _parent->getRate(); }
-private:
-	SeekableAudioStream *_parent;
-	DisposeAfterUse::Flag _disposeAfterUse;
-
-	uint _loops;
-	Timestamp _pos;
-	Timestamp _loopStart, _loopEnd;
-
-	bool _done;
-};
-
-
-/**
- * A SubSeekableAudioStream provides access to a SeekableAudioStream
- * just in the range [start, end).
- * The same caveats apply to SubSeekableAudioStream as do to SeekableAudioStream.
- *
- * Manipulating the parent stream directly /will/ mess up a substream.
- */
-class SubSeekableAudioStream : public SeekableAudioStream {
-public:
-	/**
-	 * Creates a new SubSeekableAudioStream.
-	 *
-	 * @param parent          parent stream object.
-	 * @param start           Start time.
-	 * @param end             End time.
-	 * @param disposeAfterUse Whether the parent stream object should be destroyed on destruction of the SubSeekableAudioStream.
-	 */
-	SubSeekableAudioStream(SeekableAudioStream *parent, const Timestamp start, const Timestamp end, DisposeAfterUse::Flag disposeAfterUse = DisposeAfterUse::YES);
-	~SubSeekableAudioStream();
-
-	int readBuffer(int16 *buffer, const int numSamples);
-
-	bool isStereo() const { return _parent->isStereo(); }
-
-	int getRate() const { return _parent->getRate(); }
-
-	bool endOfData() const { return (_pos >= _length) || _parent->endOfStream(); }
-
-	bool seek(const Timestamp &where);
-
-	Timestamp getLength() const { return _length; }
-private:
-	SeekableAudioStream *_parent;
-	DisposeAfterUse::Flag _disposeAfterUse;
-
-	const Timestamp _start;
-	const Timestamp _length;
-	Timestamp _pos;
-};
-
-class QueuingAudioStream : public Audio::AudioStream {
-public:
-
-	/**
-	 * Queue an audio stream for playback. This stream plays all queued
-	 * streams, in the order they were queued. If disposeAfterUse is set to
-	 * DisposeAfterUse::YES, then the queued stream is deleted after all data
-	 * contained in it has been played.
-	 */
-	virtual void queueAudioStream(Audio::AudioStream *audStream,
-	                              DisposeAfterUse::Flag disposeAfterUse = DisposeAfterUse::YES) = 0;
-
-	/**
-	 * Queue a block of raw audio data for playback. This stream plays all
-	 * queued block, in the order they were queued. If disposeAfterUse is set
-	 * to DisposeAfterUse::YES, then the queued block is released using free()
-	 * after all data contained in it has been played.
-	 *
-	 * @note Make sure to allocate the data block with malloc(), not with new[].
-	 *
-	 * @param data             pointer to the audio data block
-	 * @param size             length of the audio data block
-	 * @param disposeAfterUse  if equal to DisposeAfterUse::YES, the block is released using free() after use.
-	 * @param flags            a bit-ORed combination of RawFlags describing the audio data format
-	 */
-	void queueBuffer(byte *data, uint32 size, DisposeAfterUse::Flag disposeAfterUse, byte flags);
-
-	/**
-	 * Mark this stream as finished. That is, signal that no further data
-	 * will be queued to it. Only after this has been done can this
-	 * stream ever 'end'.
-	 */
-	virtual void finish() = 0;
-
-	/**
-	 * Return the number of streams still queued for playback (including
-	 * the currently playing stream).
-	 */
-	virtual uint32 numQueuedStreams() const = 0;
-};
-
-/**
- * Factory function for an QueuingAudioStream.
- */
-QueuingAudioStream *makeQueuingAudioStream(int rate, bool stereo);
-
-/**
- * Converts a point in time to a precise sample offset
- * with the given parameters.
- *
- * @param where    Point in time.
- * @param rate     Rate of the stream.
- * @param isStereo Is the stream a stereo stream?
- */
-Timestamp convertTimeToStreamPos(const Timestamp &where, int rate, bool isStereo);
-
-} // End of namespace Audio
-
-#endif
diff --git a/sound/decoders/adpcm.cpp b/sound/decoders/adpcm.cpp
deleted file mode 100644
index 4b5e6a5b84..0000000000
--- a/sound/decoders/adpcm.cpp
+++ /dev/null
@@ -1,851 +0,0 @@
-/* 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 "sound/decoders/adpcm.h"
-#include "sound/audiostream.h"
-
-
-namespace Audio {
-
-class ADPCMStream : public RewindableAudioStream {
-protected:
-	Common::SeekableReadStream *_stream;
-	const DisposeAfterUse::Flag _disposeAfterUse;
-	const int32 _startpos;
-	const int32 _endpos;
-	const int _channels;
-	const uint32 _blockAlign;
-	uint32 _blockPos[2];
-	const int _rate;
-
-	struct {
-		// OKI/IMA
-		struct {
-			int32 last;
-			int32 stepIndex;
-		} ima_ch[2];
-	} _status;
-
-	virtual void reset();
-	int16 stepAdjust(byte);
-
-public:
-	ADPCMStream(Common::SeekableReadStream *stream, DisposeAfterUse::Flag disposeAfterUse, uint32 size, int rate, int channels, uint32 blockAlign);
-	~ADPCMStream();
-
-	virtual bool endOfData() const { return (_stream->eos() || _stream->pos() >= _endpos); }
-	virtual bool isStereo() const	{ return _channels == 2; }
-	virtual int getRate() const	{ return _rate; }
-
-	virtual bool rewind();
-};
-
-// Routines to convert 12 bit linear samples to the
-// Dialogic or Oki ADPCM coding format aka VOX.
-// See also <http://www.comptek.ru/telephony/tnotes/tt1-13.html>
-//
-// IMA ADPCM support is based on
-//   <http://wiki.multimedia.cx/index.php?title=IMA_ADPCM>
-//
-// In addition, also MS IMA ADPCM is supported. See
-//   <http://wiki.multimedia.cx/index.php?title=Microsoft_IMA_ADPCM>.
-
-ADPCMStream::ADPCMStream(Common::SeekableReadStream *stream, DisposeAfterUse::Flag disposeAfterUse, uint32 size, int rate, int channels, uint32 blockAlign)
-	: _stream(stream),
-		_disposeAfterUse(disposeAfterUse),
-		_startpos(stream->pos()),
-		_endpos(_startpos + size),
-		_channels(channels),
-		_blockAlign(blockAlign),
-		_rate(rate) {
-
-	reset();
-}
-
-ADPCMStream::~ADPCMStream() {
-	if (_disposeAfterUse == DisposeAfterUse::YES)
-		delete _stream;
-}
-
-void ADPCMStream::reset() {
-	memset(&_status, 0, sizeof(_status));
-	_blockPos[0] = _blockPos[1] = _blockAlign; // To make sure first header is read
-}
-
-bool ADPCMStream::rewind() {
-	// TODO: Error checking.
-	reset();
-	_stream->seek(_startpos);
-	return true;
-}
-
-
-#pragma mark -
-
-
-class Oki_ADPCMStream : public ADPCMStream {
-public:
-	Oki_ADPCMStream(Common::SeekableReadStream *stream, DisposeAfterUse::Flag disposeAfterUse, uint32 size, int rate, int channels, uint32 blockAlign)
-		: ADPCMStream(stream, disposeAfterUse, size, rate, channels, blockAlign) {}
-
-	virtual int readBuffer(int16 *buffer, const int numSamples);
-
-protected:
-	int16 decodeOKI(byte);
-};
-
-int Oki_ADPCMStream::readBuffer(int16 *buffer, const int numSamples) {
-	int samples;
-	byte data;
-
-	assert(numSamples % 2 == 0);
-
-	for (samples = 0; samples < numSamples && !_stream->eos() && _stream->pos() < _endpos; samples += 2) {
-		data = _stream->readByte();
-		buffer[samples] = decodeOKI((data >> 4) & 0x0f);
-		buffer[samples + 1] = decodeOKI(data & 0x0f);
-	}
-	return samples;
-}
-
-static const int16 okiStepSize[49] = {
-	   16,   17,   19,   21,   23,   25,   28,   31,
-	   34,   37,   41,   45,   50,   55,   60,   66,
-	   73,   80,   88,   97,  107,  118,  130,  143,
-	  157,  173,  190,  209,  230,  253,  279,  307,
-	  337,  371,  408,  449,  494,  544,  598,  658,
-	  724,  796,  876,  963, 1060, 1166, 1282, 1411,
-	 1552
-};
-
-// Decode Linear to ADPCM
-int16 Oki_ADPCMStream::decodeOKI(byte code) {
-	int16 diff, E, samp;
-
-	E = (2 * (code & 0x7) + 1) * okiStepSize[_status.ima_ch[0].stepIndex] / 8;
-	diff = (code & 0x08) ? -E : E;
-	samp = _status.ima_ch[0].last + diff;
-	// Clip the values to +/- 2^11 (supposed to be 12 bits)
-	samp = CLIP<int16>(samp, -2048, 2047);
-
-	_status.ima_ch[0].last = samp;
-	_status.ima_ch[0].stepIndex += stepAdjust(code);
-	_status.ima_ch[0].stepIndex = CLIP<int32>(_status.ima_ch[0].stepIndex, 0, ARRAYSIZE(okiStepSize) - 1);
-
-	// * 16 effectively converts 12-bit input to 16-bit output
-	return samp * 16;
-}
-
-
-#pragma mark -
-
-
-class Ima_ADPCMStream : public ADPCMStream {
-protected:
-	int16 decodeIMA(byte code, int channel = 0); // Default to using the left channel/using one channel
-
-public:
-	Ima_ADPCMStream(Common::SeekableReadStream *stream, DisposeAfterUse::Flag disposeAfterUse, uint32 size, int rate, int channels, uint32 blockAlign)
-		: ADPCMStream(stream, disposeAfterUse, size, rate, channels, blockAlign) {
-		memset(&_status, 0, sizeof(_status));
-	}
-
-	virtual int readBuffer(int16 *buffer, const int numSamples);
-};
-
-int Ima_ADPCMStream::readBuffer(int16 *buffer, const int numSamples) {
-	int samples;
-	byte data;
-
-	assert(numSamples % 2 == 0);
-
-	for (samples = 0; samples < numSamples && !_stream->eos() && _stream->pos() < _endpos; samples += 2) {
-		data = _stream->readByte();
-		buffer[samples] = decodeIMA((data >> 4) & 0x0f);
-		buffer[samples + 1] = decodeIMA(data & 0x0f, _channels == 2 ? 1 : 0);
-	}
-	return samples;
-}
-
-#pragma mark -
-
-
-class Apple_ADPCMStream : public Ima_ADPCMStream {
-protected:
-	// Apple QuickTime IMA ADPCM
-	int32 _streamPos[2];
-	int16 _buffer[2][2];
-	uint8 _chunkPos[2];
-
-	void reset() {
-		Ima_ADPCMStream::reset();
-		_chunkPos[0] = 0;
-		_chunkPos[1] = 0;
-		_streamPos[0] = 0;
-		_streamPos[1] = _blockAlign;
-	}
-
-public:
-	Apple_ADPCMStream(Common::SeekableReadStream *stream, DisposeAfterUse::Flag disposeAfterUse, uint32 size, int rate, int channels, uint32 blockAlign)
-		: Ima_ADPCMStream(stream, disposeAfterUse, size, rate, channels, blockAlign) {
-		_chunkPos[0] = 0;
-		_chunkPos[1] = 0;
-		_streamPos[0] = 0;
-		_streamPos[1] = _blockAlign;
-	}
-
-	virtual int readBuffer(int16 *buffer, const int numSamples);
-
-};
-
-int Apple_ADPCMStream::readBuffer(int16 *buffer, const int numSamples) {
-	// Need to write at least one samples per channel
-	assert((numSamples % _channels) == 0);
-
-	// Current sample positions
-	int samples[2] = { 0, 0};
-
-	// Number of samples per channel
-	int chanSamples = numSamples / _channels;
-
-	for (int i = 0; i < _channels; i++) {
-		_stream->seek(_streamPos[i]);
-
-		while ((samples[i] < chanSamples) &&
-		       // Last byte read and a new one needed
-		       !((_stream->eos() || (_stream->pos() >= _endpos)) && (_chunkPos[i] == 0))) {
-
-			if (_blockPos[i] == _blockAlign) {
-				// 2 byte header per block
-				uint16 temp = _stream->readUint16BE();
-
-				// First 9 bits are the upper bits of the predictor
-				_status.ima_ch[i].last      = (int16) (temp & 0xFF80);
-				// Lower 7 bits are the step index
-				_status.ima_ch[i].stepIndex =          temp & 0x007F;
-
-				// Clip the step index
-				_status.ima_ch[i].stepIndex = CLIP<int32>(_status.ima_ch[i].stepIndex, 0, 88);
-
-				_blockPos[i] = 2;
-			}
-
-			if (_chunkPos[i] == 0) {
-				// Decode data
-				byte data = _stream->readByte();
-				_buffer[i][0] = decodeIMA(data &  0x0F, i);
-				_buffer[i][1] = decodeIMA(data >>    4, i);
-			}
-
-			// The original is interleaved block-wise, we want it sample-wise
-			buffer[_channels * samples[i] + i] = _buffer[i][_chunkPos[i]];
-
-			if (++_chunkPos[i] > 1) {
-				// We're about to decode the next byte, so advance the block position
-				_chunkPos[i] = 0;
-				_blockPos[i]++;
-			}
-
-			samples[i]++;
-
-			if (_channels == 2)
-				if (_blockPos[i] == _blockAlign)
-					// We're at the end of the block.
-					// Since the channels are interleaved, skip the next block
-					_stream->skip(MIN<uint32>(_blockAlign, _endpos - _stream->pos()));
-
-			_streamPos[i] = _stream->pos();
-		}
-	}
-
-	return samples[0] + samples[1];
-}
-
-#pragma mark -
-
-
-class MSIma_ADPCMStream : public Ima_ADPCMStream {
-public:
-	MSIma_ADPCMStream(Common::SeekableReadStream *stream, DisposeAfterUse::Flag disposeAfterUse, uint32 size, int rate, int channels, uint32 blockAlign, bool invertSamples = false)
-		: Ima_ADPCMStream(stream, disposeAfterUse, size, rate, channels, blockAlign), _invertSamples(invertSamples) {
-		if (blockAlign == 0)
-			error("ADPCMStream(): blockAlign isn't specified for MS IMA ADPCM");
-	}
-
-	virtual int readBuffer(int16 *buffer, const int numSamples) {
-		if (_channels == 1)
-			return readBufferMSIMA1(buffer, numSamples);
-		else
-			return readBufferMSIMA2(buffer, numSamples);
-	}
-
-	int readBufferMSIMA1(int16 *buffer, const int numSamples);
-	int readBufferMSIMA2(int16 *buffer, const int numSamples);
-
-private:
-	bool _invertSamples;    // Some implementations invert the way samples are decoded
-};
-
-int MSIma_ADPCMStream::readBufferMSIMA1(int16 *buffer, const int numSamples) {
-	int samples = 0;
-	byte data;
-
-	assert(numSamples % 2 == 0);
-
-	while (samples < numSamples && !_stream->eos() && _stream->pos() < _endpos) {
-		if (_blockPos[0] == _blockAlign) {
-			// read block header
-			_status.ima_ch[0].last = _stream->readSint16LE();
-			_status.ima_ch[0].stepIndex = _stream->readSint16LE();
-			_blockPos[0] = 4;
-		}
-
-		for (; samples < numSamples && _blockPos[0] < _blockAlign && !_stream->eos() && _stream->pos() < _endpos; samples += 2) {
-			data = _stream->readByte();
-			_blockPos[0]++;
-			buffer[samples] = decodeIMA(_invertSamples ? (data >> 4) & 0x0f : data & 0x0f);
-			buffer[samples + 1] = decodeIMA(_invertSamples ? data & 0x0f : (data >> 4) & 0x0f);
-		}
-	}
-	return samples;
-}
-
-
-// Microsoft as usual tries to implement it differently. This method
-// is used for stereo data.
-int MSIma_ADPCMStream::readBufferMSIMA2(int16 *buffer, const int numSamples) {
-	int samples;
-	uint32 data;
-	int nibble;
-	byte k;
-
-	// TODO: Currently this implementation only supports
-	// reading a multiple of 16 samples at once. We might
-	// consider changing that so it could read an arbitrary
-	// sample pair count.
-	assert(numSamples % 16 == 0);
-
-	for (samples = 0; samples < numSamples && !_stream->eos() && _stream->pos() < _endpos;) {
-		for (int channel = 0; channel < 2; channel++) {
-			data = _stream->readUint32LE();
-
-			for (nibble = 0; nibble < 8; nibble++) {
-				k = ((data & 0xf0000000) >> 28);
-				buffer[samples + channel + nibble * 2] = decodeIMA(k);
-				data <<= 4;
-			}
-		}
-		samples += 16;
-	}
-	return samples;
-}
-
-
-#pragma mark -
-
-
-static const int MSADPCMAdaptCoeff1[] = {
-	256, 512, 0, 192, 240, 460, 392
-};
-
-static const int MSADPCMAdaptCoeff2[] = {
-	0, -256, 0, 64, 0, -208, -232
-};
-
-static const int MSADPCMAdaptationTable[] = {
-	230, 230, 230, 230, 307, 409, 512, 614,
-	768, 614, 512, 409, 307, 230, 230, 230
-};
-
-
-class MS_ADPCMStream : public ADPCMStream {
-protected:
-	struct ADPCMChannelStatus {
-		byte predictor;
-		int16 delta;
-		int16 coeff1;
-		int16 coeff2;
-		int16 sample1;
-		int16 sample2;
-	};
-
-	struct {
-		// MS ADPCM
-		ADPCMChannelStatus ch[2];
-	} _status;
-
-	void reset() {
-		ADPCMStream::reset();
-		memset(&_status, 0, sizeof(_status));
-	}
-
-public:
-	MS_ADPCMStream(Common::SeekableReadStream *stream, DisposeAfterUse::Flag disposeAfterUse, uint32 size, int rate, int channels, uint32 blockAlign)
-		: ADPCMStream(stream, disposeAfterUse, size, rate, channels, blockAlign) {
-		if (blockAlign == 0)
-			error("MS_ADPCMStream(): blockAlign isn't specified for MS ADPCM");
-		memset(&_status, 0, sizeof(_status));
-	}
-
-	virtual int readBuffer(int16 *buffer, const int numSamples);
-
-protected:
-	int16 decodeMS(ADPCMChannelStatus *c, byte);
-};
-
-int16 MS_ADPCMStream::decodeMS(ADPCMChannelStatus *c, byte code) {
-	int32 predictor;
-
-	predictor = (((c->sample1) * (c->coeff1)) + ((c->sample2) * (c->coeff2))) / 256;
-	predictor += (signed)((code & 0x08) ? (code - 0x10) : (code)) * c->delta;
-
-	predictor = CLIP<int32>(predictor, -32768, 32767);
-
-	c->sample2 = c->sample1;
-	c->sample1 = predictor;
-	c->delta = (MSADPCMAdaptationTable[(int)code] * c->delta) >> 8;
-
-	if (c->delta < 16)
-		c->delta = 16;
-
-	return (int16)predictor;
-}
-
-int MS_ADPCMStream::readBuffer(int16 *buffer, const int numSamples) {
-	int samples;
-	byte data;
-	int i = 0;
-
-	samples = 0;
-
-	while (samples < numSamples && !_stream->eos() && _stream->pos() < _endpos) {
-		if (_blockPos[0] == _blockAlign) {
-			// read block header
-			for (i = 0; i < _channels; i++) {
-				_status.ch[i].predictor = CLIP(_stream->readByte(), (byte)0, (byte)6);
-				_status.ch[i].coeff1 = MSADPCMAdaptCoeff1[_status.ch[i].predictor];
-				_status.ch[i].coeff2 = MSADPCMAdaptCoeff2[_status.ch[i].predictor];
-			}
-
-			for (i = 0; i < _channels; i++)
-				_status.ch[i].delta = _stream->readSint16LE();
-
-			for (i = 0; i < _channels; i++)
-				_status.ch[i].sample1 = _stream->readSint16LE();
-
-			for (i = 0; i < _channels; i++)
-				buffer[samples++] = _status.ch[i].sample2 = _stream->readSint16LE();
-
-			for (i = 0; i < _channels; i++)
-				buffer[samples++] = _status.ch[i].sample1;
-
-			_blockPos[0] = _channels * 7;
-		}
-
-		for (; samples < numSamples && _blockPos[0] < _blockAlign && !_stream->eos() && _stream->pos() < _endpos; samples += 2) {
-			data = _stream->readByte();
-			_blockPos[0]++;
-			buffer[samples] = decodeMS(&_status.ch[0], (data >> 4) & 0x0f);
-			buffer[samples + 1] = decodeMS(&_status.ch[_channels - 1], data & 0x0f);
-		}
-	}
-
-	return samples;
-}
-
-
-
-#pragma mark -
-
-
-class Tinsel_ADPCMStream : public ADPCMStream {
-protected:
-	struct {
-		// Tinsel
-		double predictor;
-		double K0, K1;
-		double d0, d1;
-	} _status;
-
-	void reset() {
-		ADPCMStream::reset();
-		memset(&_status, 0, sizeof(_status));
-	}
-
-	int16 decodeTinsel(int16, double);
-	void readBufferTinselHeader();
-
-public:
-	Tinsel_ADPCMStream(Common::SeekableReadStream *stream, DisposeAfterUse::Flag disposeAfterUse, uint32 size, int rate, int channels, uint32 blockAlign)
-		: ADPCMStream(stream, disposeAfterUse, size, rate, channels, blockAlign) {
-
-		if (blockAlign == 0)
-			error("Tinsel_ADPCMStream(): blockAlign isn't specified");
-
-		if (channels != 1)
-			error("Tinsel_ADPCMStream(): Tinsel ADPCM only supports mono");
-
-		memset(&_status, 0, sizeof(_status));
-	}
-
-};
-
-static const double TinselFilterTable[4][2] = {
-	{0, 0 },
-	{0.9375, 0},
-	{1.796875, -0.8125},
-	{1.53125, -0.859375}
-};
-
-void Tinsel_ADPCMStream::readBufferTinselHeader() {
-	uint8 start = _stream->readByte();
-	uint8 filterVal = (start & 0xC0) >> 6;
-
-	if ((start & 0x20) != 0) {
-		//Lower 6 bit are negative
-
-		// Negate
-		start = ~(start | 0xC0) + 1;
-
-		_status.predictor = 1 << start;
-	} else {
-		// Lower 6 bit are positive
-
-		// Truncate
-		start &= 0x1F;
-
-		_status.predictor = ((double) 1.0) / (1 << start);
-	}
-
-	_status.K0 = TinselFilterTable[filterVal][0];
-	_status.K1 = TinselFilterTable[filterVal][1];
-}
-
-int16 Tinsel_ADPCMStream::decodeTinsel(int16 code, double eVal) {
-	double sample;
-
-	sample = (double) code;
-	sample *= eVal * _status.predictor;
-	sample += (_status.d0 * _status.K0) + (_status.d1 * _status.K1);
-
-	_status.d1 = _status.d0;
-	_status.d0 = sample;
-
-	return (int16) CLIP<double>(sample, -32768.0, 32767.0);
-}
-
-class Tinsel4_ADPCMStream : public Tinsel_ADPCMStream {
-public:
-	Tinsel4_ADPCMStream(Common::SeekableReadStream *stream, DisposeAfterUse::Flag disposeAfterUse, uint32 size, int rate, int channels, uint32 blockAlign)
-		: Tinsel_ADPCMStream(stream, disposeAfterUse, size, rate, channels, blockAlign) {}
-
-	virtual int readBuffer(int16 *buffer, const int numSamples);
-};
-
-int Tinsel4_ADPCMStream::readBuffer(int16 *buffer, const int numSamples) {
-	int samples;
-	uint16 data;
-	const double eVal = 1.142822265;
-
-	samples = 0;
-
-	assert(numSamples % 2 == 0);
-
-	while (samples < numSamples && !_stream->eos() && _stream->pos() < _endpos) {
-		if (_blockPos[0] == _blockAlign) {
-			readBufferTinselHeader();
-			_blockPos[0] = 0;
-		}
-
-		for (; samples < numSamples && _blockPos[0] < _blockAlign && !_stream->eos() && _stream->pos() < _endpos; samples += 2, _blockPos[0]++) {
-			// Read 1 byte = 8 bits = two 4 bit blocks
-			data = _stream->readByte();
-			buffer[samples] = decodeTinsel((data << 8) & 0xF000, eVal);
-			buffer[samples+1] = decodeTinsel((data << 12) & 0xF000, eVal);
-		}
-	}
-
-	return samples;
-}
-
-class Tinsel6_ADPCMStream : public Tinsel_ADPCMStream {
-protected:
-	uint8 _chunkPos;
-	uint16 _chunkData;
-
-	void reset() {
-		ADPCMStream::reset();
-		_chunkPos = 0;
-		_chunkData = 0;
-	}
-
-public:
-	Tinsel6_ADPCMStream(Common::SeekableReadStream *stream, DisposeAfterUse::Flag disposeAfterUse, uint32 size, int rate, int channels, uint32 blockAlign)
-		: Tinsel_ADPCMStream(stream, disposeAfterUse, size, rate, channels, blockAlign) {
-		_chunkPos = 0;
-		_chunkData = 0;
-	}
-
-	virtual int readBuffer(int16 *buffer, const int numSamples);
-};
-
-int Tinsel6_ADPCMStream::readBuffer(int16 *buffer, const int numSamples) {
-	int samples;
-	const double eVal = 1.032226562;
-
-	samples = 0;
-
-	while (samples < numSamples && !_stream->eos() && _stream->pos() < _endpos) {
-		if (_blockPos[0] == _blockAlign) {
-			readBufferTinselHeader();
-			_blockPos[0] = 0;
-			_chunkPos = 0;
-		}
-
-		for (; samples < numSamples && _blockPos[0] < _blockAlign && !_stream->eos() && _stream->pos() < _endpos; samples++, _chunkPos = (_chunkPos + 1) % 4) {
-
-			switch (_chunkPos) {
-			case 0:
-				_chunkData = _stream->readByte();
-				buffer[samples] = decodeTinsel((_chunkData << 8) & 0xFC00, eVal);
-				break;
-			case 1:
-				_chunkData = (_chunkData << 8) | (_stream->readByte());
-				buffer[samples] = decodeTinsel((_chunkData << 6) & 0xFC00, eVal);
-				_blockPos[0]++;
-				break;
-			case 2:
-				_chunkData = (_chunkData << 8) | (_stream->readByte());
-				buffer[samples] = decodeTinsel((_chunkData << 4) & 0xFC00, eVal);
-				_blockPos[0]++;
-				break;
-			case 3:
-				_chunkData = (_chunkData << 8);
-				buffer[samples] = decodeTinsel((_chunkData << 2) & 0xFC00, eVal);
-				_blockPos[0]++;
-				break;
-			}
-
-		}
-
-	}
-
-	return samples;
-}
-
-class Tinsel8_ADPCMStream : public Tinsel_ADPCMStream {
-public:
-	Tinsel8_ADPCMStream(Common::SeekableReadStream *stream, DisposeAfterUse::Flag disposeAfterUse, uint32 size, int rate, int channels, uint32 blockAlign)
-		: Tinsel_ADPCMStream(stream, disposeAfterUse, size, rate, channels, blockAlign) {}
-
-	virtual int readBuffer(int16 *buffer, const int numSamples);
-};
-
-int Tinsel8_ADPCMStream::readBuffer(int16 *buffer, const int numSamples) {
-	int samples;
-	byte data;
-	const double eVal = 1.007843258;
-
-	samples = 0;
-
-	while (samples < numSamples && !_stream->eos() && _stream->pos() < _endpos) {
-		if (_blockPos[0] == _blockAlign) {
-			readBufferTinselHeader();
-			_blockPos[0] = 0;
-		}
-
-		for (; samples < numSamples && _blockPos[0] < _blockAlign && !_stream->eos() && _stream->pos() < _endpos; samples++, _blockPos[0]++) {
-			// Read 1 byte = 8 bits = one 8 bit block
-			data = _stream->readByte();
-			buffer[samples] = decodeTinsel(data << 8, eVal);
-		}
-	}
-
-	return samples;
-}
-
-
-#pragma mark -
-
-// Duck DK3 IMA ADPCM Decoder
-// Based on FFmpeg's decoder and http://wiki.multimedia.cx/index.php?title=Duck_DK3_IMA_ADPCM
-
-class DK3_ADPCMStream : public Ima_ADPCMStream {
-protected:
-
-	void reset() {
-		Ima_ADPCMStream::reset();
-		_topNibble = false;
-	}
-
-public:
-	DK3_ADPCMStream(Common::SeekableReadStream *stream, DisposeAfterUse::Flag disposeAfterUse, uint32 size, int rate, int channels, uint32 blockAlign)
-		: Ima_ADPCMStream(stream, disposeAfterUse, size, rate, channels, blockAlign) {
-
-		// DK3 only works as a stereo stream
-		assert(channels == 2);
-		_topNibble = false;
-	}
-
-	virtual int readBuffer(int16 *buffer, const int numSamples);
-
-private:
-	byte _nibble, _lastByte;
-	bool _topNibble;
-};
-
-#define DK3_READ_NIBBLE() \
-do { \
-	if (_topNibble) { \
-		_nibble = _lastByte >> 4; \
-		_topNibble = false; \
-	} else { \
-		if (_stream->pos() >= _endpos) \
-			break; \
-		if ((_stream->pos() % _blockAlign) == 0) \
-			continue; \
-		_lastByte = _stream->readByte(); \
-		_nibble = _lastByte & 0xf; \
-		_topNibble = true; \
-	} \
-} while (0)
-		
-
-int DK3_ADPCMStream::readBuffer(int16 *buffer, const int numSamples) {
-	int samples = 0;
-
-	assert((numSamples % 4) == 0);
-
-	while (samples < numSamples && !_stream->eos() && _stream->pos() < _endpos) {	
-		if ((_stream->pos() % _blockAlign) == 0) {
-			_stream->readUint16LE(); // Unknown
-			uint16 rate = _stream->readUint16LE(); // Copy of rate
-			_stream->skip(6); // Unknown
-			// Get predictor for both sum/diff channels
-			_status.ima_ch[0].last = _stream->readSint16LE();
-			_status.ima_ch[1].last = _stream->readSint16LE();
-			// Get index for both sum/diff channels
-			_status.ima_ch[0].stepIndex = _stream->readByte();
-			_status.ima_ch[1].stepIndex = _stream->readByte();
-
-			if (_stream->eos())
-				break;
-
-			// Sanity check
-			assert(rate == getRate());
-		}
-
-		DK3_READ_NIBBLE();
-		decodeIMA(_nibble, 0);
-
-		DK3_READ_NIBBLE();
-		decodeIMA(_nibble, 1);
-
-		buffer[samples++] = _status.ima_ch[0].last + _status.ima_ch[1].last;
-		buffer[samples++] = _status.ima_ch[0].last - _status.ima_ch[1].last;
-
-		DK3_READ_NIBBLE();
-		decodeIMA(_nibble, 0);
-
-		buffer[samples++] = _status.ima_ch[0].last + _status.ima_ch[1].last;
-		buffer[samples++] = _status.ima_ch[0].last - _status.ima_ch[1].last;
-	}
-
-	return samples;
-}
-
-
-#pragma mark -
-
-
-// adjust the step for use on the next sample.
-int16 ADPCMStream::stepAdjust(byte code) {
-	static const int16 adjusts[] = {-1, -1, -1, -1, 2, 4, 6, 8};
-
-	return adjusts[code & 0x07];
-}
-
-static const uint16 imaStepTable[89] = {
-		7,    8,    9,   10,   11,   12,   13,   14,
-	   16,   17,   19,   21,   23,   25,   28,   31,
-	   34,   37,   41,   45,   50,   55,   60,   66,
-	   73,   80,   88,   97,  107,  118,  130,  143,
-	  157,  173,  190,  209,  230,  253,  279,  307,
-	  337,  371,  408,  449,  494,  544,  598,  658,
-	  724,  796,  876,  963, 1060, 1166, 1282, 1411,
-	 1552, 1707, 1878, 2066, 2272, 2499, 2749, 3024,
-	 3327, 3660, 4026, 4428, 4871, 5358, 5894, 6484,
-	 7132, 7845, 8630, 9493,10442,11487,12635,13899,
-	15289,16818,18500,20350,22385,24623,27086,29794,
-	32767
-};
-
-int16 Ima_ADPCMStream::decodeIMA(byte code, int channel) {
-	int32 E = (2 * (code & 0x7) + 1) * imaStepTable[_status.ima_ch[channel].stepIndex] / 8;
-	int32 diff = (code & 0x08) ? -E : E;
-	int32 samp = CLIP<int32>(_status.ima_ch[channel].last + diff, -32768, 32767);
-
-	_status.ima_ch[channel].last = samp;
-	_status.ima_ch[channel].stepIndex += stepAdjust(code);
-	_status.ima_ch[channel].stepIndex = CLIP<int32>(_status.ima_ch[channel].stepIndex, 0, ARRAYSIZE(imaStepTable) - 1);
-
-	return samp;
-}
-
-RewindableAudioStream *makeADPCMStream(Common::SeekableReadStream *stream, DisposeAfterUse::Flag disposeAfterUse, uint32 size, typesADPCM type, int rate, int channels, uint32 blockAlign) {
-	// If size is 0, report the entire size of the stream
-	if (!size)
-		size = stream->size();
-
-	switch (type) {
-	case kADPCMOki:
-		return new Oki_ADPCMStream(stream, disposeAfterUse, size, rate, channels, blockAlign);
-	case kADPCMMSIma:
-		return new MSIma_ADPCMStream(stream, disposeAfterUse, size, rate, channels, blockAlign);
-	case kADPCMMSImaLastExpress:
-		return new MSIma_ADPCMStream(stream, disposeAfterUse, size, rate, channels, blockAlign, true);
-	case kADPCMMS:
-		return new MS_ADPCMStream(stream, disposeAfterUse, size, rate, channels, blockAlign);
-	case kADPCMTinsel4:
-		return new Tinsel4_ADPCMStream(stream, disposeAfterUse, size, rate, channels, blockAlign);
-	case kADPCMTinsel6:
-		return new Tinsel6_ADPCMStream(stream, disposeAfterUse, size, rate, channels, blockAlign);
-	case kADPCMTinsel8:
-		return new Tinsel8_ADPCMStream(stream, disposeAfterUse, size, rate, channels, blockAlign);
-	case kADPCMIma:
-		return new Ima_ADPCMStream(stream, disposeAfterUse, size, rate, channels, blockAlign);
-	case kADPCMApple:
-		return new Apple_ADPCMStream(stream, disposeAfterUse, size, rate, channels, blockAlign);
-	case kADPCMDK3:
-		return new DK3_ADPCMStream(stream, disposeAfterUse, size, rate, channels, blockAlign);
-	default:
-		error("Unsupported ADPCM encoding");
-		break;
-	}
-}
-
-} // End of namespace Audio
diff --git a/sound/decoders/adpcm.h b/sound/decoders/adpcm.h
deleted file mode 100644
index 38ec870a27..0000000000
--- a/sound/decoders/adpcm.h
+++ /dev/null
@@ -1,90 +0,0 @@
-/* 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
- *  - lastexpress
- *  - mohawk
- *  - saga
- *  - scumm
- *  - tinsel
- */
-
-#ifndef SOUND_ADPCM_H
-#define SOUND_ADPCM_H
-
-#include "common/scummsys.h"
-#include "common/stream.h"
-
-
-namespace Audio {
-
-class AudioStream;
-class RewindableAudioStream;
-
-// There are several types of ADPCM encoding, only some are supported here
-// For all the different encodings, refer to:
-// http://wiki.multimedia.cx/index.php?title=Category:ADPCM_Audio_Codecs
-// Usually, if the audio stream we're trying to play has the FourCC header
-// string intact, it's easy to discern which encoding is used
-enum typesADPCM {
-	kADPCMOki,                 // Dialogic/Oki ADPCM (aka VOX)
-	kADPCMMSIma,               // Microsoft IMA ADPCM
-	kADPCMMSImaLastExpress,    // Microsoft IMA ADPCM (with inverted samples)
-	kADPCMMS,                  // Microsoft ADPCM
-	kADPCMTinsel4,             // 4-bit ADPCM used by the Tinsel engine
-	kADPCMTinsel6,             // 6-bit ADPCM used by the Tinsel engine
-	kADPCMTinsel8,             // 8-bit ADPCM used by the Tinsel engine
-	kADPCMIma,                 // Standard IMA ADPCM
-	kADPCMApple,               // Apple QuickTime IMA ADPCM
-	kADPCMDK3                  // Duck DK3 IMA ADPCM
-};
-
-/**
- * Takes an input stream containing ADPCM compressed sound data and creates
- * an RewindableAudioStream from that.
- *
- * @param stream            the SeekableReadStream from which to read the ADPCM data
- * @param disposeAfterUse   whether to delete the stream after use
- * @param size              how many bytes to read from the stream (0 = all)
- * @param type              the compression type used
- * @param rate              the sampling rate
- * @param channels          the number of channels
- * @param blockAlign        block alignment ???
- * @return   a new RewindableAudioStream, or NULL, if an error occurred
- */
-RewindableAudioStream *makeADPCMStream(
-    Common::SeekableReadStream *stream,
-    DisposeAfterUse::Flag disposeAfterUse,
-    uint32 size, typesADPCM type,
-    int rate = 22050,
-    int channels = 2,
-    uint32 blockAlign = 0);
-
-} // End of namespace Audio
-
-#endif
diff --git a/sound/decoders/aiff.cpp b/sound/decoders/aiff.cpp
deleted file mode 100644
index ce8c6ad32c..0000000000
--- a/sound/decoders/aiff.cpp
+++ /dev/null
@@ -1,186 +0,0 @@
-/* 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.
- *
- * $URL$
- * $Id$
- *
- */
-
-/*
- * The code in this file is based on information found at
- * http://www.borg.com/~jglatt/tech/aiff.htm
- *
- * We currently only implement uncompressed AIFF. If we ever need AIFF-C, SoX
- * (http://sox.sourceforge.net) may be a good place to start from.
- */
-
-#include "common/endian.h"
-#include "common/util.h"
-#include "common/stream.h"
-
-#include "sound/decoders/aiff.h"
-#include "sound/audiostream.h"
-#include "sound/mixer.h"
-#include "sound/decoders/raw.h"
-
-namespace Audio {
-
-uint32 readExtended(Common::SeekableReadStream &stream) {
-	// The sample rate is stored as an "80 bit IEEE Standard 754 floating
-	// point number (Standard Apple Numeric Environment [SANE] data type
-	// Extended).
-
-	byte buf[10];
-	uint32 mantissa;
-	uint32 last = 0;
-	byte exp;
-
-	stream.read(buf, 10);
-	mantissa = READ_BE_UINT32(buf + 2);
-	exp = 30 - buf[1];
-
-	while (exp--) {
-		last = mantissa;
-		mantissa >>= 1;
-	}
-
-	if (last & 0x00000001)
-		mantissa++;
-
-	return mantissa;
-}
-
-bool loadAIFFFromStream(Common::SeekableReadStream &stream, int &size, int &rate, byte &flags) {
-	byte buf[4];
-
-	stream.read(buf, 4);
-	if (memcmp(buf, "FORM", 4) != 0) {
-		warning("loadAIFFFromStream: No 'FORM' header");
-		return false;
-	}
-
-	stream.readUint32BE();
-
-	// This could be AIFC, but we don't handle that case.
-
-	stream.read(buf, 4);
-	if (memcmp(buf, "AIFF", 4) != 0) {
-		warning("loadAIFFFromStream: No 'AIFF' header");
-		return false;
-	}
-
-	// From here on, we only care about the COMM and SSND chunks, which are
-	// the only required chunks.
-
-	bool foundCOMM = false;
-	bool foundSSND = false;
-
-	uint16 numChannels = 0, bitsPerSample = 0;
-	uint32 numSampleFrames = 0, offset = 0, blockSize = 0, soundOffset = 0;
-
-	while (!(foundCOMM && foundSSND) && !stream.err() && !stream.eos()) {
-		uint32 length, pos;
-
-		stream.read(buf, 4);
-		length = stream.readUint32BE();
-		pos = stream.pos();
-
-		if (memcmp(buf, "COMM", 4) == 0) {
-			foundCOMM = true;
-			numChannels = stream.readUint16BE();
-			numSampleFrames = stream.readUint32BE();
-			bitsPerSample = stream.readUint16BE();
-			rate = readExtended(stream);
-			size = numSampleFrames * numChannels * (bitsPerSample / 8);
-		} else if (memcmp(buf, "SSND", 4) == 0) {
-			foundSSND = true;
-			offset = stream.readUint32BE();
-			blockSize = stream.readUint32BE();
-			soundOffset = stream.pos();
-		}
-
-		stream.seek(pos + length);
-	}
-
-	if (!foundCOMM) {
-		warning("loadAIFFFromStream: Cound not find 'COMM' chunk");
-		return false;
-	}
-
-	if (!foundSSND) {
-		warning("loadAIFFFromStream: Cound not find 'SSND' chunk");
-		return false;
-	}
-
-	// We only implement a subset of the AIFF standard.
-
-	if (numChannels < 1 || numChannels > 2) {
-		warning("loadAIFFFromStream: Only 1 or 2 channels are supported, not %d", numChannels);
-		return false;
-	}
-
-	if (bitsPerSample != 8 && bitsPerSample != 16) {
-		warning("loadAIFFFromStream: Only 8 or 16 bits per sample are supported, not %d", bitsPerSample);
-		return false;
-	}
-
-	if (offset != 0 || blockSize != 0) {
-		warning("loadAIFFFromStream: Block-aligned data is not supported");
-		return false;
-	}
-
-	// Samples are always signed, and big endian.
-
-	flags = 0;
-	if (bitsPerSample == 16)
-		flags |= Audio::FLAG_16BITS;
-	if (numChannels == 2)
-		flags |= Audio::FLAG_STEREO;
-
-	stream.seek(soundOffset);
-
-	// Stream now points at the sample data
-
-	return true;
-}
-
-SeekableAudioStream *makeAIFFStream(Common::SeekableReadStream *stream,
-	DisposeAfterUse::Flag disposeAfterUse) {
-	int size, rate;
-	byte *data, flags;
-
-	if (!loadAIFFFromStream(*stream, size, rate, flags)) {
-		if (disposeAfterUse == DisposeAfterUse::YES)
-			delete stream;
-		return 0;
-	}
-
-	data = (byte *)malloc(size);
-	assert(data);
-	stream->read(data, size);
-
-	if (disposeAfterUse == DisposeAfterUse::YES)
-		delete stream;
-
-	// Since we allocated our own buffer for the data, we must specify DisposeAfterUse::YES.
-	return makeRawStream(data, size, rate, flags);
-}
-
-} // End of namespace Audio
diff --git a/sound/decoders/aiff.h b/sound/decoders/aiff.h
deleted file mode 100644
index 06c56ecd38..0000000000
--- a/sound/decoders/aiff.h
+++ /dev/null
@@ -1,71 +0,0 @@
-/* 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:
- *  - saga
- *  - sci
- *  - sword1
- */
-
-#ifndef SOUND_AIFF_H
-#define SOUND_AIFF_H
-
-#include "common/scummsys.h"
-#include "common/types.h"
-
-namespace Common { class SeekableReadStream; }
-
-namespace Audio {
-
-class SeekableAudioStream;
-
-/**
- * Try to load an AIFF from the given seekable stream. Returns true if
- * successful. In that case, the stream's seek position will be set to the
- * start of the audio data, and size, rate and flags contain information
- * necessary for playback. Currently this function only supports uncompressed
- * raw PCM data as well as IMA ADPCM.
- */
-extern bool loadAIFFFromStream(Common::SeekableReadStream &stream, int &size, int &rate, byte &flags);
-
-/**
- * Try to load an AIFF from the given seekable stream and create an AudioStream
- * from that data.
- *
- * This function uses loadAIFFFromStream() internally.
- *
- * @param stream			the SeekableReadStream from which to read the AIFF data
- * @param disposeAfterUse	whether to delete the stream after use
- * @return	a new SeekableAudioStream, or NULL, if an error occurred
- */
-SeekableAudioStream *makeAIFFStream(
-	Common::SeekableReadStream *stream,
-	DisposeAfterUse::Flag disposeAfterUse);
-
-} // End of namespace Audio
-
-#endif
diff --git a/sound/decoders/flac.cpp b/sound/decoders/flac.cpp
deleted file mode 100644
index 080141f224..0000000000
--- a/sound/decoders/flac.cpp
+++ /dev/null
@@ -1,745 +0,0 @@
-/* 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.
- *
- * $URL$
- * $Id$
- *
- */
-
-// Disable symbol overrides for FILE as that is used in FLAC headers
-#define FORBIDDEN_SYMBOL_EXCEPTION_FILE
-
-#include "sound/decoders/flac.h"
-
-#ifdef USE_FLAC
-
-#include "common/debug.h"
-#include "common/stream.h"
-#include "common/util.h"
-
-#include "sound/audiostream.h"
-
-#define FLAC__NO_DLL // that MS-magic gave me headaches - just link the library you like
-#include <FLAC/export.h>
-
-
-// check if we have FLAC >= 1.1.3; LEGACY_FLAC code can be removed once FLAC-1.1.3 propagates everywhere
-#if !defined(FLAC_API_VERSION_CURRENT) || FLAC_API_VERSION_CURRENT < 8
-#define LEGACY_FLAC
-#else
-#undef LEGACY_FLAC
-#endif
-
-
-#ifdef LEGACY_FLAC
-
-// Before FLAC 1.1.3, we needed to use the stream decoder API.
-#include <FLAC/seekable_stream_decoder.h>
-typedef uint FLAC_size_t;
-
-#else
-
-// With FLAC 1.1.3, the stream decoder API was merged into the regular
-// stream API. In order to stay compatible with older FLAC versions, we
-// simply add some typedefs and #ifdefs to map between the old and new API.
-// We use the typedefs (instead of only #defines) in order to somewhat
-// improve the readability of the code.
-
-#include <FLAC/stream_decoder.h>
-typedef size_t FLAC_size_t;
-// Add aliases for the old names
-typedef FLAC__StreamDecoderState FLAC__SeekableStreamDecoderState;
-typedef FLAC__StreamDecoderReadStatus FLAC__SeekableStreamDecoderReadStatus;
-typedef FLAC__StreamDecoderSeekStatus FLAC__SeekableStreamDecoderSeekStatus;
-typedef FLAC__StreamDecoderTellStatus FLAC__SeekableStreamDecoderTellStatus;
-typedef FLAC__StreamDecoderLengthStatus FLAC__SeekableStreamDecoderLengthStatus;
-typedef FLAC__StreamDecoder FLAC__SeekableStreamDecoder;
-
-#endif
-
-
-namespace Audio {
-
-#pragma mark -
-#pragma mark --- FLAC stream ---
-#pragma mark -
-
-static const uint MAX_OUTPUT_CHANNELS = 2;
-
-
-class FLACStream : public SeekableAudioStream {
-protected:
-	Common::SeekableReadStream *_inStream;
-	bool _disposeAfterUse;
-
-	::FLAC__SeekableStreamDecoder *_decoder;
-
-	/** Header of the stream */
-	FLAC__StreamMetadata_StreamInfo _streaminfo;
-
-	/** index + 1(!) of the last sample to be played */
-	FLAC__uint64 _lastSample;
-
-	/** total play time */
-	Timestamp _length;
-
-	/** true if the last sample was decoded from the FLAC-API - there might still be data in the buffer */
-	bool _lastSampleWritten;
-
-	typedef int16 SampleType;
-	enum { BUFTYPE_BITS = 16 };
-
-	enum {
-		// Maximal buffer size. According to the FLAC format specification, the  block size is
-		// a 16 bit value (in fact it seems the maximal block size is 32768, but we play it safe).
-		BUFFER_SIZE = 65536
-	};
-
-	struct {
-		SampleType bufData[BUFFER_SIZE];
-		SampleType *bufReadPos;
-		uint bufFill;
-	} _sampleCache;
-
-	SampleType *_outBuffer;
-	uint _requestedSamples;
-
-	typedef void (*PFCONVERTBUFFERS)(SampleType*, const FLAC__int32*[], uint, const uint, const uint8);
-	PFCONVERTBUFFERS _methodConvertBuffers;
-
-
-public:
-	FLACStream(Common::SeekableReadStream *inStream, bool dispose);
-	virtual ~FLACStream();
-
-	int readBuffer(int16 *buffer, const int numSamples);
-
-	bool isStereo() const { return _streaminfo.channels >= 2; }
-	int getRate() const { return _streaminfo.sample_rate; }
-	bool endOfData() const {
-		// End of data is reached if there either is no valid stream data available,
-		// or if we reached the last sample and completely emptied the sample cache.
-		return _streaminfo.channels == 0 || (_lastSampleWritten && _sampleCache.bufFill == 0);
-	}
-
-	bool seek(const Timestamp &where);
-	Timestamp getLength() const { return _length; }
-
-	bool isStreamDecoderReady() const { return getStreamDecoderState() == FLAC__STREAM_DECODER_SEARCH_FOR_FRAME_SYNC; }
-protected:
-	uint getChannels() const { return MIN<uint>(_streaminfo.channels, MAX_OUTPUT_CHANNELS); }
-
-	bool allocateBuffer(uint minSamples);
-
-	inline FLAC__StreamDecoderState getStreamDecoderState() const;
-
-	inline bool processSingleBlock();
-	inline bool processUntilEndOfMetadata();
-	bool seekAbsolute(FLAC__uint64 sample);
-
-	inline ::FLAC__SeekableStreamDecoderReadStatus callbackRead(FLAC__byte buffer[], FLAC_size_t *bytes);
-	inline ::FLAC__SeekableStreamDecoderSeekStatus callbackSeek(FLAC__uint64 absoluteByteOffset);
-	inline ::FLAC__SeekableStreamDecoderTellStatus callbackTell(FLAC__uint64 *absoluteByteOffset);
-	inline ::FLAC__SeekableStreamDecoderLengthStatus callbackLength(FLAC__uint64 *streamLength);
-	inline bool callbackEOF();
-	inline ::FLAC__StreamDecoderWriteStatus callbackWrite(const ::FLAC__Frame *frame, const FLAC__int32 * const buffer[]);
-	inline void callbackMetadata(const ::FLAC__StreamMetadata *metadata);
-	inline void callbackError(::FLAC__StreamDecoderErrorStatus status);
-
-private:
-	static ::FLAC__SeekableStreamDecoderReadStatus callWrapRead(const ::FLAC__SeekableStreamDecoder *decoder, FLAC__byte buffer[], FLAC_size_t *bytes, void *clientData);
-	static ::FLAC__SeekableStreamDecoderSeekStatus callWrapSeek(const ::FLAC__SeekableStreamDecoder *decoder, FLAC__uint64 absoluteByteOffset, void *clientData);
-	static ::FLAC__SeekableStreamDecoderTellStatus callWrapTell(const ::FLAC__SeekableStreamDecoder *decoder, FLAC__uint64 *absoluteByteOffset, void *clientData);
-	static ::FLAC__SeekableStreamDecoderLengthStatus callWrapLength(const ::FLAC__SeekableStreamDecoder *decoder, FLAC__uint64 *streamLength, void *clientData);
-	static FLAC__bool callWrapEOF(const ::FLAC__SeekableStreamDecoder *decoder, void *clientData);
-	static ::FLAC__StreamDecoderWriteStatus callWrapWrite(const ::FLAC__SeekableStreamDecoder *decoder, const ::FLAC__Frame *frame, const FLAC__int32 * const buffer[], void *clientData);
-	static void callWrapMetadata(const ::FLAC__SeekableStreamDecoder *decoder, const ::FLAC__StreamMetadata *metadata, void *clientData);
-	static void callWrapError(const ::FLAC__SeekableStreamDecoder *decoder, ::FLAC__StreamDecoderErrorStatus status, void *clientData);
-
-	void setBestConvertBufferMethod();
-	static void convertBuffersGeneric(SampleType* bufDestination, const FLAC__int32 *inChannels[], uint numSamples, const uint numChannels, const uint8 numBits);
-	static void convertBuffersStereoNS(SampleType* bufDestination, const FLAC__int32 *inChannels[], uint numSamples, const uint numChannels, const uint8 numBits);
-	static void convertBuffersStereo8Bit(SampleType* bufDestination, const FLAC__int32 *inChannels[], uint numSamples, const uint numChannels, const uint8 numBits);
-	static void convertBuffersMonoNS(SampleType* bufDestination, const FLAC__int32 *inChannels[], uint numSamples, const uint numChannels, const uint8 numBits);
-	static void convertBuffersMono8Bit(SampleType* bufDestination, const FLAC__int32 *inChannels[], uint numSamples, const uint numChannels, const uint8 numBits);
-};
-
-FLACStream::FLACStream(Common::SeekableReadStream *inStream, bool dispose)
-#ifdef LEGACY_FLAC
-			:	_decoder(::FLAC__seekable_stream_decoder_new()),
-#else
-			:	_decoder(::FLAC__stream_decoder_new()),
-#endif
-		_inStream(inStream),
-		_disposeAfterUse(dispose),
-		_length(0, 1000), _lastSample(0),
-		_outBuffer(NULL), _requestedSamples(0), _lastSampleWritten(false),
-		_methodConvertBuffers(&FLACStream::convertBuffersGeneric)
-{
-	assert(_inStream);
-	memset(&_streaminfo, 0, sizeof(_streaminfo));
-
-	_sampleCache.bufReadPos = NULL;
-	_sampleCache.bufFill = 0;
-
-	_methodConvertBuffers = &FLACStream::convertBuffersGeneric;
-
-	bool success;
-#ifdef LEGACY_FLAC
-	::FLAC__seekable_stream_decoder_set_read_callback(_decoder, &FLACStream::callWrapRead);
-	::FLAC__seekable_stream_decoder_set_seek_callback(_decoder, &FLACStream::callWrapSeek);
-	::FLAC__seekable_stream_decoder_set_tell_callback(_decoder, &FLACStream::callWrapTell);
-	::FLAC__seekable_stream_decoder_set_length_callback(_decoder, &FLACStream::callWrapLength);
-	::FLAC__seekable_stream_decoder_set_eof_callback(_decoder, &FLACStream::callWrapEOF);
-	::FLAC__seekable_stream_decoder_set_write_callback(_decoder, &FLACStream::callWrapWrite);
-	::FLAC__seekable_stream_decoder_set_metadata_callback(_decoder, &FLACStream::callWrapMetadata);
-	::FLAC__seekable_stream_decoder_set_error_callback(_decoder, &FLACStream::callWrapError);
-	::FLAC__seekable_stream_decoder_set_client_data(_decoder, (void*)this);
-
-	success = (::FLAC__seekable_stream_decoder_init(_decoder) == FLAC__SEEKABLE_STREAM_DECODER_OK);
-#else
-	success = (::FLAC__stream_decoder_init_stream(
-		_decoder,
-		&FLACStream::callWrapRead,
-		&FLACStream::callWrapSeek,
-		&FLACStream::callWrapTell,
-		&FLACStream::callWrapLength,
-		&FLACStream::callWrapEOF,
-		&FLACStream::callWrapWrite,
-		&FLACStream::callWrapMetadata,
-		&FLACStream::callWrapError,
-		(void*)this
-	) == FLAC__STREAM_DECODER_INIT_STATUS_OK);
-#endif
-	if (success) {
-		if (processUntilEndOfMetadata() && _streaminfo.channels > 0) {
-			_lastSample = _streaminfo.total_samples + 1;
-			_length = Timestamp(0, _lastSample - 1, getRate());
-			return; // no error occurred
-		}
-	}
-
-	warning("FLACStream: could not create audio stream");
-}
-
-FLACStream::~FLACStream() {
-	if (_decoder != NULL) {
-#ifdef LEGACY_FLAC
-		(void) ::FLAC__seekable_stream_decoder_finish(_decoder);
-		::FLAC__seekable_stream_decoder_delete(_decoder);
-#else
-		(void) ::FLAC__stream_decoder_finish(_decoder);
-		::FLAC__stream_decoder_delete(_decoder);
-#endif
-	}
-	if (_disposeAfterUse)
-		delete _inStream;
-}
-
-inline FLAC__StreamDecoderState FLACStream::getStreamDecoderState() const {
-	assert(_decoder != NULL);
-#ifdef LEGACY_FLAC
-	return ::FLAC__seekable_stream_decoder_get_stream_decoder_state(_decoder);
-#else
-	return ::FLAC__stream_decoder_get_state(_decoder);
-#endif
-}
-
-inline bool FLACStream::processSingleBlock() {
-	assert(_decoder != NULL);
-#ifdef LEGACY_FLAC
-	return 0 != ::FLAC__seekable_stream_decoder_process_single(_decoder);
-#else
-	return 0 != ::FLAC__stream_decoder_process_single(_decoder);
-#endif
-}
-
-inline bool FLACStream::processUntilEndOfMetadata() {
-	assert(_decoder != NULL);
-#ifdef LEGACY_FLAC
-	return 0 != ::FLAC__seekable_stream_decoder_process_until_end_of_metadata(_decoder);
-#else
-	return 0 != ::FLAC__stream_decoder_process_until_end_of_metadata(_decoder);
-#endif
-}
-
-bool FLACStream::seekAbsolute(FLAC__uint64 sample) {
-	assert(_decoder != NULL);
-#ifdef LEGACY_FLAC
-	const bool result = (0 != ::FLAC__seekable_stream_decoder_seek_absolute(_decoder, sample));
-#else
-	const bool result = (0 != ::FLAC__stream_decoder_seek_absolute(_decoder, sample));
-#endif
-	if (result) {
-		_lastSampleWritten = (_lastSample != 0 && sample >= _lastSample); // only set if we are SURE
-	}
-	return result;
-}
-
-bool FLACStream::seek(const Timestamp &where) {
-	_sampleCache.bufFill = 0;
-	_sampleCache.bufReadPos = NULL;
-	// FLAC uses the sample pair number, thus we always use "false" for the isStereo parameter
-	// of the convertTimeToStreamPos helper.
-	return seekAbsolute((FLAC__uint64)convertTimeToStreamPos(where, getRate(), false).totalNumberOfFrames());
-}
-
-int FLACStream::readBuffer(int16 *buffer, const int numSamples) {
-	const uint numChannels = getChannels();
-
-	if (numChannels == 0) {
-		warning("FLACStream: Stream not successfully initialised, cant playback");
-		return -1; // streaminfo wasnt read!
-	}
-
-	assert(numSamples % numChannels == 0); // must be multiple of channels!
-	assert(buffer != NULL);
-	assert(_outBuffer == NULL);
-	assert(_requestedSamples == 0);
-
-	_outBuffer = buffer;
-	_requestedSamples = numSamples;
-
-	// If there is still data in our buffer from the last time around,
-	// copy that first.
-	if (_sampleCache.bufFill > 0) {
-		assert(_sampleCache.bufReadPos >= _sampleCache.bufData);
-		assert(_sampleCache.bufFill % numChannels == 0);
-
-		const uint copySamples = MIN((uint)numSamples, _sampleCache.bufFill);
-		memcpy(buffer, _sampleCache.bufReadPos, copySamples*sizeof(buffer[0]));
-
-		_outBuffer = buffer + copySamples;
-		_requestedSamples = numSamples - copySamples;
-		_sampleCache.bufReadPos += copySamples;
-		_sampleCache.bufFill -= copySamples;
-	}
-
-	bool decoderOk = true;
-
-	FLAC__StreamDecoderState state = getStreamDecoderState();
-
-	// Keep poking FLAC to process more samples until we completely satisfied the request
-	// respectively until we run out of data.
-	while (!_lastSampleWritten && _requestedSamples > 0 && state == FLAC__STREAM_DECODER_SEARCH_FOR_FRAME_SYNC) {
-		assert(_sampleCache.bufFill == 0);
-		assert(_requestedSamples % numChannels == 0);
-		processSingleBlock();
-		state = getStreamDecoderState();
-
-		if (state == FLAC__STREAM_DECODER_END_OF_STREAM)
-			_lastSampleWritten = true;
-	}
-
-	// Error handling
-	switch (state) {
-	case FLAC__STREAM_DECODER_END_OF_STREAM:
-		_lastSampleWritten = true;
-		break;
-	case FLAC__STREAM_DECODER_SEARCH_FOR_FRAME_SYNC:
-		break;
-	default:
-		decoderOk = false;
-		warning("FLACStream: An error occurred while decoding. DecoderState is: %s",
-			FLAC__StreamDecoderStateString[getStreamDecoderState()]);
-	}
-
-	// Compute how many samples we actually produced
-	const int samples = (int)(_outBuffer - buffer);
-	assert(samples % numChannels == 0);
-
-	_outBuffer = NULL; // basically unnecessary, only for the purpose of the asserts
-	_requestedSamples = 0; // basically unnecessary, only for the purpose of the asserts
-
-	return decoderOk ? samples : -1;
-}
-
-inline ::FLAC__SeekableStreamDecoderReadStatus FLACStream::callbackRead(FLAC__byte buffer[], FLAC_size_t *bytes) {
-	if (*bytes == 0) {
-#ifdef LEGACY_FLAC
-		return FLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_ERROR; /* abort to avoid a deadlock */
-#else
-		return FLAC__STREAM_DECODER_READ_STATUS_ABORT; /* abort to avoid a deadlock */
-#endif
-	}
-
-	const uint32 bytesRead = _inStream->read(buffer, *bytes);
-
-	if (bytesRead == 0) {
-#ifdef LEGACY_FLAC
-		return _inStream->eos() ? FLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_OK : FLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_ERROR;
-#else
-		return _inStream->eos() ? FLAC__STREAM_DECODER_READ_STATUS_END_OF_STREAM : FLAC__STREAM_DECODER_READ_STATUS_ABORT;
-#endif
-	}
-
-	*bytes = static_cast<uint>(bytesRead);
-#ifdef LEGACY_FLAC
-	return FLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_OK;
-#else
-	return FLAC__STREAM_DECODER_READ_STATUS_CONTINUE;
-#endif
-}
-
-void FLACStream::setBestConvertBufferMethod() {
-	PFCONVERTBUFFERS tempMethod = &FLACStream::convertBuffersGeneric;
-
-	const uint numChannels = getChannels();
-	const uint8 numBits = (uint8)_streaminfo.bits_per_sample;
-
-	assert(numChannels >= 1);
-	assert(numBits >= 4 && numBits <=32);
-
-	if (numChannels == 1) {
-		if (numBits == 8)
-			tempMethod = &FLACStream::convertBuffersMono8Bit;
-		if (numBits == BUFTYPE_BITS)
-			tempMethod = &FLACStream::convertBuffersMonoNS;
-	} else if (numChannels == 2) {
-		if (numBits == 8)
-			tempMethod = &FLACStream::convertBuffersStereo8Bit;
-		if (numBits == BUFTYPE_BITS)
-			tempMethod = &FLACStream::convertBuffersStereoNS;
-	} /* else ... */
-
-	_methodConvertBuffers = tempMethod;
-}
-
-// 1 channel, no scaling
-void FLACStream::convertBuffersMonoNS(SampleType* bufDestination, const FLAC__int32 *inChannels[], uint numSamples, const uint numChannels, const uint8 numBits) {
-	assert(numChannels == 1);
-	assert(numBits == BUFTYPE_BITS);
-
-	FLAC__int32 const* inChannel1 = inChannels[0];
-
-	while (numSamples >= 4) {
-		bufDestination[0] = static_cast<SampleType>(inChannel1[0]);
-		bufDestination[1] = static_cast<SampleType>(inChannel1[1]);
-		bufDestination[2] = static_cast<SampleType>(inChannel1[2]);
-		bufDestination[3] = static_cast<SampleType>(inChannel1[3]);
-		bufDestination += 4;
-		inChannel1 += 4;
-		numSamples -= 4;
-	}
-
-	for (; numSamples > 0; --numSamples) {
-		*bufDestination++ = static_cast<SampleType>(*inChannel1++);
-	}
-
-	inChannels[0] = inChannel1;
-	assert(numSamples == 0); // dint copy too many samples
-}
-
-// 1 channel, scaling from 8Bit
-void FLACStream::convertBuffersMono8Bit(SampleType* bufDestination, const FLAC__int32 *inChannels[], uint numSamples, const uint numChannels, const uint8 numBits) {
-	assert(numChannels == 1);
-	assert(numBits == 8);
-	assert(8 < BUFTYPE_BITS);
-
-	FLAC__int32 const* inChannel1 = inChannels[0];
-
-	while (numSamples >= 4) {
-		bufDestination[0] = static_cast<SampleType>(inChannel1[0]) << (BUFTYPE_BITS - 8);
-		bufDestination[1] = static_cast<SampleType>(inChannel1[1]) << (BUFTYPE_BITS - 8);
-		bufDestination[2] = static_cast<SampleType>(inChannel1[2]) << (BUFTYPE_BITS - 8);
-		bufDestination[3] = static_cast<SampleType>(inChannel1[3]) << (BUFTYPE_BITS - 8);
-		bufDestination += 4;
-		inChannel1 += 4;
-		numSamples -= 4;
-	}
-
-	for (; numSamples > 0; --numSamples) {
-		*bufDestination++ = static_cast<SampleType>(*inChannel1++) << (BUFTYPE_BITS - 8);
-	}
-
-	inChannels[0] = inChannel1;
-	assert(numSamples == 0); // dint copy too many samples
-}
-
-// 2 channels, no scaling
-void FLACStream::convertBuffersStereoNS(SampleType* bufDestination, const FLAC__int32 *inChannels[], uint numSamples, const uint numChannels, const uint8 numBits) {
-	assert(numChannels == 2);
-	assert(numBits == BUFTYPE_BITS);
-	assert(numSamples % 2 == 0); // must be integral multiply of channels
-
-
-	FLAC__int32 const* inChannel1 = inChannels[0];	// Left Channel
-	FLAC__int32 const* inChannel2 = inChannels[1];	// Right Channel
-
-	while (numSamples >= 2*2) {
-		bufDestination[0] = static_cast<SampleType>(inChannel1[0]);
-		bufDestination[1] = static_cast<SampleType>(inChannel2[0]);
-		bufDestination[2] = static_cast<SampleType>(inChannel1[1]);
-		bufDestination[3] = static_cast<SampleType>(inChannel2[1]);
-		bufDestination += 2 * 2;
-		inChannel1 += 2;
-		inChannel2 += 2;
-		numSamples -= 2 * 2;
-	}
-
-	while (numSamples > 0) {
-		bufDestination[0] = static_cast<SampleType>(*inChannel1++);
-		bufDestination[1] = static_cast<SampleType>(*inChannel2++);
-		bufDestination += 2;
-		numSamples -= 2;
-	}
-
-	inChannels[0] = inChannel1;
-	inChannels[1] = inChannel2;
-	assert(numSamples == 0); // dint copy too many samples
-}
-
-// 2 channels, scaling from 8Bit
-void FLACStream::convertBuffersStereo8Bit(SampleType* bufDestination, const FLAC__int32 *inChannels[], uint numSamples, const uint numChannels, const uint8 numBits) {
-	assert(numChannels == 2);
-	assert(numBits == 8);
-	assert(numSamples % 2 == 0); // must be integral multiply of channels
-	assert(8 < BUFTYPE_BITS);
-
-	FLAC__int32 const* inChannel1 = inChannels[0];	// Left Channel
-	FLAC__int32 const* inChannel2 = inChannels[1];	// Right Channel
-
-	while (numSamples >= 2*2) {
-		bufDestination[0] = static_cast<SampleType>(inChannel1[0]) << (BUFTYPE_BITS - 8);
-		bufDestination[1] = static_cast<SampleType>(inChannel2[0]) << (BUFTYPE_BITS - 8);
-		bufDestination[2] = static_cast<SampleType>(inChannel1[1]) << (BUFTYPE_BITS - 8);
-		bufDestination[3] = static_cast<SampleType>(inChannel2[1]) << (BUFTYPE_BITS - 8);
-		bufDestination += 2 * 2;
-		inChannel1 += 2;
-		inChannel2 += 2;
-		numSamples -= 2 * 2;
-	}
-
-	while (numSamples > 0) {
-		bufDestination[0] = static_cast<SampleType>(*inChannel1++) << (BUFTYPE_BITS - 8);
-		bufDestination[1] = static_cast<SampleType>(*inChannel2++) << (BUFTYPE_BITS - 8);
-		bufDestination += 2;
-		numSamples -= 2;
-	}
-
-	inChannels[0] = inChannel1;
-	inChannels[1] = inChannel2;
-	assert(numSamples == 0); // dint copy too many samples
-}
-
-// all Purpose-conversion - slowest of em all
-void FLACStream::convertBuffersGeneric(SampleType* bufDestination, const FLAC__int32 *inChannels[], uint numSamples, const uint numChannels, const uint8 numBits) {
-	assert(numSamples % numChannels == 0); // must be integral multiply of channels
-
-	if (numBits < BUFTYPE_BITS) {
-		const uint8 kPower = (uint8)(BUFTYPE_BITS - numBits);
-
-		for (; numSamples > 0; numSamples -= numChannels) {
-			for (uint i = 0; i < numChannels; ++i)
-				*bufDestination++ = static_cast<SampleType>(*(inChannels[i]++)) << kPower;
-		}
-	} else if (numBits > BUFTYPE_BITS) {
-		const uint8 kPower = (uint8)(numBits - BUFTYPE_BITS);
-
-		for (; numSamples > 0; numSamples -= numChannels) {
-			for (uint i = 0; i < numChannels; ++i)
-				*bufDestination++ = static_cast<SampleType>(*(inChannels[i]++) >> kPower);
-		}
-	} else {
-		for (; numSamples > 0; numSamples -= numChannels) {
-			for (uint i = 0; i < numChannels; ++i)
-				*bufDestination++ = static_cast<SampleType>(*(inChannels[i]++));
-		}
-	}
-
-	assert(numSamples == 0); // dint copy too many samples
-}
-
-inline ::FLAC__StreamDecoderWriteStatus FLACStream::callbackWrite(const ::FLAC__Frame *frame, const FLAC__int32 * const buffer[]) {
-	assert(frame->header.channels == _streaminfo.channels);
-	assert(frame->header.sample_rate == _streaminfo.sample_rate);
-	assert(frame->header.bits_per_sample == _streaminfo.bits_per_sample);
-	assert(frame->header.number_type == FLAC__FRAME_NUMBER_TYPE_SAMPLE_NUMBER || _streaminfo.min_blocksize == _streaminfo.max_blocksize);
-
-	// We require that either the sample cache is empty, or that no samples were requested
-	assert(_sampleCache.bufFill == 0 || _requestedSamples == 0);
-
-	uint numSamples = frame->header.blocksize;
-	const uint numChannels = getChannels();
-	const uint8 numBits = (uint8)_streaminfo.bits_per_sample;
-
-	assert(_requestedSamples % numChannels == 0); // must be integral multiply of channels
-
-	const FLAC__uint64 firstSampleNumber = (frame->header.number_type == FLAC__FRAME_NUMBER_TYPE_SAMPLE_NUMBER) ?
-		frame->header.number.sample_number : (static_cast<FLAC__uint64>(frame->header.number.frame_number)) * _streaminfo.max_blocksize;
-
-	// Check whether we are about to reach beyond the last sample we are supposed to play.
-	if (_lastSample != 0 && firstSampleNumber + numSamples >= _lastSample) {
-		numSamples = (uint)(firstSampleNumber >= _lastSample ? 0 : _lastSample - firstSampleNumber);
-		_lastSampleWritten = true;
-	}
-
-	// The value in _requestedSamples counts raw samples, so if there are more than one
-	// channel, we have to multiply the number of available sample "pairs" by numChannels
-	numSamples *= numChannels;
-
-	const FLAC__int32 *inChannels[MAX_OUTPUT_CHANNELS];
-	for (uint i = 0; i < numChannels; ++i)
-		inChannels[i] = buffer[i];
-
-	// write the incoming samples directly into the buffer provided to us by the mixer
-	if (_requestedSamples > 0) {
-		assert(_requestedSamples % numChannels == 0);
-		assert(_outBuffer != NULL);
-
-		// Copy & convert the available samples (limited both by how many we have available, and
-		// by how many are actually needed).
-		const uint copySamples = MIN(_requestedSamples, numSamples);
-		(*_methodConvertBuffers)(_outBuffer, inChannels, copySamples, numChannels, numBits);
-
-		_requestedSamples -= copySamples;
-		numSamples -= copySamples;
-		_outBuffer += copySamples;
-	}
-
-	// Write all remaining samples (i.e. those which didn't fit into the mixer buffer)
-	// into the sample cache.
-	if (_sampleCache.bufFill == 0)
-		_sampleCache.bufReadPos = _sampleCache.bufData;
-	const uint cacheSpace = (_sampleCache.bufData + BUFFER_SIZE) - (_sampleCache.bufReadPos + _sampleCache.bufFill);
-	assert(numSamples <= cacheSpace);
-	(*_methodConvertBuffers)(_sampleCache.bufReadPos + _sampleCache.bufFill, inChannels, numSamples, numChannels, numBits);
-
-	_sampleCache.bufFill += numSamples;
-
-	return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
-}
-
-inline ::FLAC__SeekableStreamDecoderSeekStatus FLACStream::callbackSeek(FLAC__uint64 absoluteByteOffset) {
-	_inStream->seek(absoluteByteOffset, SEEK_SET);
-	const bool result = (absoluteByteOffset == (FLAC__uint64)_inStream->pos());
-
-#ifdef LEGACY_FLAC
-	return result ? FLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_OK : FLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_ERROR;
-#else
-	return result ? FLAC__STREAM_DECODER_SEEK_STATUS_OK : FLAC__STREAM_DECODER_SEEK_STATUS_ERROR;
-#endif
-}
-
-inline ::FLAC__SeekableStreamDecoderTellStatus FLACStream::callbackTell(FLAC__uint64 *absoluteByteOffset) {
-	*absoluteByteOffset = static_cast<FLAC__uint64>(_inStream->pos());
-#ifdef LEGACY_FLAC
-	return FLAC__SEEKABLE_STREAM_DECODER_TELL_STATUS_OK;
-#else
-	return FLAC__STREAM_DECODER_TELL_STATUS_OK;
-#endif
-}
-
-inline ::FLAC__SeekableStreamDecoderLengthStatus FLACStream::callbackLength(FLAC__uint64 *streamLength) {
-	*streamLength = static_cast<FLAC__uint64>(_inStream->size());
-#ifdef LEGACY_FLAC
-	return FLAC__SEEKABLE_STREAM_DECODER_LENGTH_STATUS_OK;
-#else
-	return FLAC__STREAM_DECODER_LENGTH_STATUS_OK;
-#endif
-}
-
-inline bool FLACStream::callbackEOF() {
-	return _inStream->eos();
-}
-
-
-inline void FLACStream::callbackMetadata(const ::FLAC__StreamMetadata *metadata) {
-	assert(_decoder != NULL);
-	assert(metadata->type == FLAC__METADATA_TYPE_STREAMINFO); // others arent really interesting
-
-	_streaminfo = metadata->data.stream_info;
-	setBestConvertBufferMethod(); // should be set after getting stream-information. FLAC always parses the info first
-}
-inline void FLACStream::callbackError(::FLAC__StreamDecoderErrorStatus status) {
-	// some of these are non-critical-Errors
-	debug(1, "FLACStream: An error occurred while decoding. DecoderState is: %s",
-			FLAC__StreamDecoderErrorStatusString[status]);
-}
-
-/* Static Callback Wrappers */
-::FLAC__SeekableStreamDecoderReadStatus FLACStream::callWrapRead(const ::FLAC__SeekableStreamDecoder *decoder, FLAC__byte buffer[], FLAC_size_t *bytes, void *clientData) {
-	FLACStream *instance = (FLACStream *)clientData;
-	assert(0 != instance);
-	return instance->callbackRead(buffer, bytes);
-}
-
-::FLAC__SeekableStreamDecoderSeekStatus FLACStream::callWrapSeek(const ::FLAC__SeekableStreamDecoder *decoder, FLAC__uint64 absoluteByteOffset, void *clientData) {
-	FLACStream *instance = (FLACStream *)clientData;
-	assert(0 != instance);
-	return instance->callbackSeek(absoluteByteOffset);
-}
-
-::FLAC__SeekableStreamDecoderTellStatus FLACStream::callWrapTell(const ::FLAC__SeekableStreamDecoder *decoder, FLAC__uint64 *absoluteByteOffset, void *clientData) {
-	FLACStream *instance = (FLACStream *)clientData;
-	assert(0 != instance);
-	return instance->callbackTell(absoluteByteOffset);
-}
-
-::FLAC__SeekableStreamDecoderLengthStatus FLACStream::callWrapLength(const ::FLAC__SeekableStreamDecoder *decoder, FLAC__uint64 *streamLength, void *clientData) {
-	FLACStream *instance = (FLACStream *)clientData;
-	assert(0 != instance);
-	return instance->callbackLength(streamLength);
-}
-
-FLAC__bool FLACStream::callWrapEOF(const ::FLAC__SeekableStreamDecoder *decoder, void *clientData) {
-	FLACStream *instance = (FLACStream *)clientData;
-	assert(0 != instance);
-	return instance->callbackEOF();
-}
-
-::FLAC__StreamDecoderWriteStatus FLACStream::callWrapWrite(const ::FLAC__SeekableStreamDecoder *decoder, const ::FLAC__Frame *frame, const FLAC__int32 * const buffer[], void *clientData) {
-	FLACStream *instance = (FLACStream *)clientData;
-	assert(0 != instance);
-	return instance->callbackWrite(frame, buffer);
-}
-
-void FLACStream::callWrapMetadata(const ::FLAC__SeekableStreamDecoder *decoder, const ::FLAC__StreamMetadata *metadata, void *clientData) {
-	FLACStream *instance = (FLACStream *)clientData;
-	assert(0 != instance);
-	instance->callbackMetadata(metadata);
-}
-
-void FLACStream::callWrapError(const ::FLAC__SeekableStreamDecoder *decoder, ::FLAC__StreamDecoderErrorStatus status, void *clientData) {
-	FLACStream *instance = (FLACStream *)clientData;
-	assert(0 != instance);
-	instance->callbackError(status);
-}
-
-
-#pragma mark -
-#pragma mark --- FLAC factory functions ---
-#pragma mark -
-
-SeekableAudioStream *makeFLACStream(
-	Common::SeekableReadStream *stream,
-	DisposeAfterUse::Flag disposeAfterUse) {
-	SeekableAudioStream *s = new FLACStream(stream, disposeAfterUse);
-	if (s && s->endOfData()) {
-		delete s;
-		return 0;
-	} else {
-		return s;
-	}
-}
-
-} // End of namespace Audio
-
-#endif // #ifdef USE_FLAC
diff --git a/sound/decoders/flac.h b/sound/decoders/flac.h
deleted file mode 100644
index 17f95ec1fb..0000000000
--- a/sound/decoders/flac.h
+++ /dev/null
@@ -1,75 +0,0 @@
-/* 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
- *  - draci
- *  - kyra
- *  - m4
- *  - queen
- *  - saga
- *  - sci
- *  - scumm
- *  - sword1
- *  - sword2
- *  - touche
- *  - tucker
- */
-
-#ifndef SOUND_FLAC_H
-#define SOUND_FLAC_H
-
-#include "common/scummsys.h"
-#include "common/types.h"
-
-#ifdef USE_FLAC
-
-namespace Common {
-	class SeekableReadStream;
-}
-
-namespace Audio {
-
-class AudioStream;
-class SeekableAudioStream;
-
-/**
- * Create a new SeekableAudioStream from the FLAC data in the given stream.
- * Allows for seeking (which is why we require a SeekableReadStream).
- *
- * @param stream            the SeekableReadStream from which to read the FLAC data
- * @param disposeAfterUse   whether to delete the stream after use
- * @return   a new SeekableAudioStream, or NULL, if an error occurred
- */
-SeekableAudioStream *makeFLACStream(
-	Common::SeekableReadStream *stream,
-	DisposeAfterUse::Flag disposeAfterUse);
-
-} // End of namespace Audio
-
-#endif // #ifdef USE_FLAC
-#endif // #ifndef SOUND_FLAC_H
diff --git a/sound/decoders/iff_sound.cpp b/sound/decoders/iff_sound.cpp
deleted file mode 100644
index 148de5b621..0000000000
--- a/sound/decoders/iff_sound.cpp
+++ /dev/null
@@ -1,130 +0,0 @@
-/* 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 "sound/decoders/iff_sound.h"
-#include "sound/audiostream.h"
-#include "sound/mixer.h"
-#include "sound/decoders/raw.h"
-#include "common/iff_container.h"
-#include "common/func.h"
-
-namespace Audio {
-
-struct Voice8Header {
-	uint32	oneShotHiSamples;
-	uint32	repeatHiSamples;
-	uint32	samplesPerHiCycle;
-	uint16	samplesPerSec;
-	byte	octaves;
-	byte	compression;
-	uint32	volume;
-
-	Voice8Header() {
-		memset(this, 0, sizeof(Voice8Header));
-	}
-
-	void load(Common::ReadStream &stream);
-};
-
-void Voice8Header::load(Common::ReadStream &stream) {
-	oneShotHiSamples = stream.readUint32BE();
-	repeatHiSamples = stream.readUint32BE();
-	samplesPerHiCycle = stream.readUint32BE();
-	samplesPerSec = stream.readUint16BE();
-	octaves = stream.readByte();
-	compression = stream.readByte();
-	volume = stream.readUint32BE();
-}
-
-
-
-struct A8SVXLoader {
-	Voice8Header _header;
-	int8 *_data;
-	uint32 _dataSize;
-
-	void load(Common::ReadStream &input) {
-		Common::IFFParser parser(&input);
-		Common::Functor1Mem< Common::IFFChunk&, bool, A8SVXLoader > c(this, &A8SVXLoader::callback);
-		parser.parse(c);
-	}
-
-	bool callback(Common::IFFChunk &chunk) {
-		switch (chunk._type) {
-		case ID_VHDR:
-			_header.load(*chunk._stream);
-			break;
-
-		case ID_BODY:
-			_dataSize = chunk._size;
-			_data = (int8*)malloc(_dataSize);
-			assert(_data);
-			loadData(chunk._stream);
-			return true;
-		}
-
-		return false;
-	}
-
-	void loadData(Common::ReadStream *stream) {
-		switch (_header.compression) {
-		case 0:
-			stream->read(_data, _dataSize);
-			break;
-
-		case 1:
-			// implement other formats here
-			error("compressed IFF audio is not supported");
-			break;
-		}
-
-	}
-};
-
-
-AudioStream *make8SVXStream(Common::ReadStream &input, bool loop) {
-	A8SVXLoader loader;
-	loader.load(input);
-
-	SeekableAudioStream *stream = Audio::makeRawStream((byte *)loader._data, loader._dataSize, loader._header.samplesPerSec, 0);
-
-	uint32 loopStart = 0, loopEnd = 0;
-	if (loop) {
-		// the standard way to loop 8SVX audio implies use of the oneShotHiSamples and
-		// repeatHiSamples fields
-		loopStart = 0;
-		loopEnd = loader._header.oneShotHiSamples + loader._header.repeatHiSamples;
-
-		if (loopStart != loopEnd) {
-			return new SubLoopingAudioStream(stream, 0,
-					Timestamp(0, loopStart, loader._header.samplesPerSec),
-					Timestamp(0, loopEnd, loader._header.samplesPerSec));
-		}
-	}
-
-	return stream;
-}
-
-}
diff --git a/sound/decoders/iff_sound.h b/sound/decoders/iff_sound.h
deleted file mode 100644
index 4e53059380..0000000000
--- a/sound/decoders/iff_sound.h
+++ /dev/null
@@ -1,47 +0,0 @@
-/* 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:
- *  - parallaction
- */
-
-#ifndef SOUND_IFF_H
-#define SOUND_IFF_H
-
-namespace Common {
-	class ReadStream;
-}
-
-namespace Audio {
-
-class AudioStream;
-
-AudioStream *make8SVXStream(Common::ReadStream &stream, bool loop);
-
-}
-
-#endif
diff --git a/sound/decoders/mac_snd.cpp b/sound/decoders/mac_snd.cpp
deleted file mode 100644
index 48f6886bf4..0000000000
--- a/sound/decoders/mac_snd.cpp
+++ /dev/null
@@ -1,116 +0,0 @@
-/* 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.
- *
- * $URL$
- * $Id$
- *
- */
-
-/*
- * The code in this file is based on information found at
- * http://developer.apple.com/legacy/mac/library/documentation/mac/Sound/Sound-60.html#HEADING60-15
- *
- * We implement both type 1 and type 2 snd resources, but only those that are sampled
- */
-
-#include "common/util.h"
-#include "common/stream.h"
-
-#include "sound/decoders/mac_snd.h"
-#include "sound/audiostream.h"
-#include "sound/decoders/raw.h"
-
-namespace Audio {
-
-SeekableAudioStream *makeMacSndStream(Common::SeekableReadStream *stream,
-	DisposeAfterUse::Flag disposeAfterUse) {
-
-	uint16 sndType = stream->readUint16BE();
-
-	if (sndType == 1) {
-		// "normal" snd resources
-		if (stream->readUint16BE() != 1) {
-			warning("makeMacSndStream(): Unsupported data type count");
-			return 0;
-		}
-
-		if (stream->readUint16BE() != 5) {
-			// 5 == sampled
-			warning("makeMacSndStream(): Unsupported data type");
-			return 0;
-		}
-
-		stream->readUint32BE(); // initialization option
-	} else if (sndType == 2) {
-		// old HyperCard snd resources
-		stream->readUint16BE(); // reference count (unused)
-	} else {
-		warning("makeMacSndStream(): Unknown format type %d", sndType);
-		return 0;
-	}
-
-	// We really should never get this as long as we have sampled data only
-	if (stream->readUint16BE() != 1) {
-		warning("makeMacSndStream(): Unsupported command count");
-		return 0;
-	}
-
-	uint16 command = stream->readUint16BE();
-
-	// 0x8050 - soundCmd  (with dataOffsetFlag set): install a sampled sound as a voice
-	// 0x8051 - bufferCmd (with dataOffsetFlag set): play a sample sound
-	if (command != 0x8050 && command != 0x8051) {
-		warning("makeMacSndStream(): Unsupported command %04x", command);
-		return 0;
-	}
-
-	stream->readUint16BE(); // 0
-	uint32 soundHeaderOffset = stream->readUint32BE();
-
-	stream->seek(soundHeaderOffset);
-
-	uint32 soundDataOffset = stream->readUint32BE();
-	uint32 size = stream->readUint32BE();
-	uint16 rate = stream->readUint32BE() >> 16; // Really fixed point, but we only support integer rates
-	stream->readUint32BE(); // loop start
-	stream->readUint32BE(); // loop end
-	byte encoding = stream->readByte();
-	stream->readByte(); // base frequency
-
-	if (encoding != 0) {
-		// 0 == PCM
-		warning("makeMacSndStream(): Unsupported compression %d", encoding);
-		return 0;
-	}
-
-	stream->skip(soundDataOffset);
-
-	byte *data = (byte *)malloc(size);
-	assert(data);
-	stream->read(data, size);
-
-	if (disposeAfterUse == DisposeAfterUse::YES)
-		delete stream;
-
-	// Since we allocated our own buffer for the data, we must specify DisposeAfterUse::YES.
-	return makeRawStream(data, size, rate, Audio::FLAG_UNSIGNED);
-}
-
-} // End of namespace Audio
diff --git a/sound/decoders/mac_snd.h b/sound/decoders/mac_snd.h
deleted file mode 100644
index 198a61333e..0000000000
--- a/sound/decoders/mac_snd.h
+++ /dev/null
@@ -1,58 +0,0 @@
-/* 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:
- *  - sci
- */
-
-#ifndef SOUND_MAC_SND_H
-#define SOUND_MAC_SND_H
-
-#include "common/scummsys.h"
-#include "common/types.h"
-
-namespace Common { class SeekableReadStream; }
-
-namespace Audio {
-
-class SeekableAudioStream;
-
-/**
- * Try to load a Mac snd resource from the given seekable stream and create a SeekableAudioStream
- * from that data.
- *
- * @param stream			the SeekableReadStream from which to read the snd data
- * @param disposeAfterUse	whether to delete the stream after use
- * @return	a new SeekableAudioStream, or NULL, if an error occurred
- */
-SeekableAudioStream *makeMacSndStream(
-	Common::SeekableReadStream *stream,
-	DisposeAfterUse::Flag disposeAfterUse);
-
-} // End of namespace Audio
-
-#endif
diff --git a/sound/decoders/mp3.cpp b/sound/decoders/mp3.cpp
deleted file mode 100644
index e06b82a9e2..0000000000
--- a/sound/decoders/mp3.cpp
+++ /dev/null
@@ -1,375 +0,0 @@
-/* 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 "sound/decoders/mp3.h"
-
-#ifdef USE_MAD
-
-#include "common/debug.h"
-#include "common/stream.h"
-#include "common/util.h"
-
-#include "sound/audiostream.h"
-
-#include <mad.h>
-
-#if defined(__PSP__)
-	#include "backends/platform/psp/mp3.h"
-#endif
-namespace Audio {
-
-
-#pragma mark -
-#pragma mark --- MP3 (MAD) stream ---
-#pragma mark -
-
-
-class MP3Stream : public SeekableAudioStream {
-protected:
-	enum State {
-		MP3_STATE_INIT,	// Need to init the decoder
-		MP3_STATE_READY,	// ready for processing data
-		MP3_STATE_EOS		// end of data reached (may need to loop)
-	};
-
-	Common::SeekableReadStream *_inStream;
-	DisposeAfterUse::Flag _disposeAfterUse;
-
-	uint _posInFrame;
-	State _state;
-
-	Timestamp _length;
-	mad_timer_t _totalTime;
-
-	mad_stream _stream;
-	mad_frame _frame;
-	mad_synth _synth;
-
-	enum {
-		BUFFER_SIZE = 5 * 8192
-	};
-
-	// This buffer contains a slab of input data
-	byte _buf[BUFFER_SIZE + MAD_BUFFER_GUARD];
-
-public:
-	MP3Stream(Common::SeekableReadStream *inStream,
-	               DisposeAfterUse::Flag dispose);
-	~MP3Stream();
-
-	int readBuffer(int16 *buffer, const int numSamples);
-
-	bool endOfData() const		{ return _state == MP3_STATE_EOS; }
-	bool isStereo() const		{ return MAD_NCHANNELS(&_frame.header) == 2; }
-	int getRate() const			{ return _frame.header.samplerate; }
-
-	bool seek(const Timestamp &where);
-	Timestamp getLength() const { return _length; }
-protected:
-	void decodeMP3Data();
-	void readMP3Data();
-
-	void initStream();
-	void readHeader();
-	void deinitStream();
-};
-
-MP3Stream::MP3Stream(Common::SeekableReadStream *inStream, DisposeAfterUse::Flag dispose) :
-	_inStream(inStream),
-	_disposeAfterUse(dispose),
-	_posInFrame(0),
-	_state(MP3_STATE_INIT),
-	_length(0, 1000),
-	_totalTime(mad_timer_zero) {
-
-	// The MAD_BUFFER_GUARD must always contain zeros (the reason
-	// for this is that the Layer III Huffman decoder of libMAD
-	// may read a few bytes beyond the end of the input buffer).
-	memset(_buf + BUFFER_SIZE, 0, MAD_BUFFER_GUARD);
-
-	// Calculate the length of the stream
-	initStream();
-
-	while (_state != MP3_STATE_EOS)
-		readHeader();
-
-	// To rule out any invalid sample rate to be encountered here, say in case the
-	// MP3 stream is invalid, we just check the MAD error code here.
-	// We need to assure this, since else we might trigger an assertion in Timestamp
-	// (When getRate() returns 0 or a negative number to be precise).
-	// Note that we allow "MAD_ERROR_BUFLEN" as error code here, since according
-	// to mad.h it is also set on EOF.
-	if ((_stream.error == MAD_ERROR_NONE || _stream.error == MAD_ERROR_BUFLEN) && getRate() > 0)
-		_length = Timestamp(mad_timer_count(_totalTime, MAD_UNITS_MILLISECONDS), getRate());
-
-	deinitStream();
-
-	// Reinit stream
-	_state = MP3_STATE_INIT;
-
-	// Decode the first chunk of data. This is necessary so that _frame
-	// is setup and isStereo() and getRate() return correct results.
-	decodeMP3Data();
-}
-
-MP3Stream::~MP3Stream() {
-	deinitStream();
-
-	if (_disposeAfterUse == DisposeAfterUse::YES)
-		delete _inStream;
-}
-
-void MP3Stream::decodeMP3Data() {
-	do {
-		if (_state == MP3_STATE_INIT)
-			initStream();
-
-		if (_state == MP3_STATE_EOS)
-			return;
-
-		// If necessary, load more data into the stream decoder
-		if (_stream.error == MAD_ERROR_BUFLEN)
-			readMP3Data();
-
-		while (_state == MP3_STATE_READY) {
-			_stream.error = MAD_ERROR_NONE;
-
-			// Decode the next frame
-			if (mad_frame_decode(&_frame, &_stream) == -1) {
-				if (_stream.error == MAD_ERROR_BUFLEN) {
-					break; // Read more data
-				} else if (MAD_RECOVERABLE(_stream.error)) {
-					// Note: we will occasionally see MAD_ERROR_BADDATAPTR errors here.
-					// These are normal and expected (caused by our frame skipping (i.e. "seeking")
-					// code above).
-					debug(6, "MP3Stream: Recoverable error in mad_frame_decode (%s)", mad_stream_errorstr(&_stream));
-					continue;
-				} else {
-					warning("MP3Stream: Unrecoverable error in mad_frame_decode (%s)", mad_stream_errorstr(&_stream));
-					break;
-				}
-			}
-
-			// Synthesize PCM data
-			mad_synth_frame(&_synth, &_frame);
-			_posInFrame = 0;
-			break;
-		}
-	} while (_state != MP3_STATE_EOS && _stream.error == MAD_ERROR_BUFLEN);
-
-	if (_stream.error != MAD_ERROR_NONE)
-		_state = MP3_STATE_EOS;
-}
-
-void MP3Stream::readMP3Data() {
-	uint32 remaining = 0;
-
-	// Give up immediately if we already used up all data in the stream
-	if (_inStream->eos()) {
-		_state = MP3_STATE_EOS;
-		return;
-	}
-
-	if (_stream.next_frame) {
-		// If there is still data in the MAD stream, we need to preserve it.
-		// Note that we use memmove, as we are reusing the same buffer,
-		// and hence the data regions we copy from and to may overlap.
-		remaining = _stream.bufend - _stream.next_frame;
-		assert(remaining < BUFFER_SIZE);	// Paranoia check
-		memmove(_buf, _stream.next_frame, remaining);
-	}
-
-	// Try to read the next block
-	uint32 size = _inStream->read(_buf + remaining, BUFFER_SIZE - remaining);
-	if (size <= 0) {
-		_state = MP3_STATE_EOS;
-		return;
-	}
-
-	// Feed the data we just read into the stream decoder
-	_stream.error = MAD_ERROR_NONE;
-	mad_stream_buffer(&_stream, _buf, size + remaining);
-}
-
-bool MP3Stream::seek(const Timestamp &where) {
-	if (where == _length) {
-		_state = MP3_STATE_EOS;
-		return true;
-	} else if (where > _length) {
-		return false;
-	}
-
-	const uint32 time = where.msecs();
-
-	mad_timer_t destination;
-	mad_timer_set(&destination, time / 1000, time % 1000, 1000);
-
-	if (_state != MP3_STATE_READY || mad_timer_compare(destination, _totalTime) < 0)
-		initStream();
-
-	while (mad_timer_compare(destination, _totalTime) > 0 && _state != MP3_STATE_EOS)
-		readHeader();
-
-	decodeMP3Data();
-
-	return (_state != MP3_STATE_EOS);
-}
-
-void MP3Stream::initStream() {
-	if (_state != MP3_STATE_INIT)
-		deinitStream();
-
-	// Init MAD
-	mad_stream_init(&_stream);
-	mad_frame_init(&_frame);
-	mad_synth_init(&_synth);
-
-	// Reset the stream data
-	_inStream->seek(0, SEEK_SET);
-	_totalTime = mad_timer_zero;
-	_posInFrame = 0;
-
-	// Update state
-	_state = MP3_STATE_READY;
-
-	// Read the first few sample bytes
-	readMP3Data();
-}
-
-void MP3Stream::readHeader() {
-	if (_state != MP3_STATE_READY)
-		return;
-
-	// If necessary, load more data into the stream decoder
-	if (_stream.error == MAD_ERROR_BUFLEN)
-		readMP3Data();
-
-	while (_state != MP3_STATE_EOS) {
-		_stream.error = MAD_ERROR_NONE;
-
-		// Decode the next header. Note: mad_frame_decode would do this for us, too.
-		// However, for seeking we don't want to decode the full frame (else it would
-		// be far too slow). Hence we perform this explicitly in a separate step.
-		if (mad_header_decode(&_frame.header, &_stream) == -1) {
-			if (_stream.error == MAD_ERROR_BUFLEN) {
-				readMP3Data();  // Read more data
-				continue;
-			} else if (MAD_RECOVERABLE(_stream.error)) {
-				debug(6, "MP3Stream: Recoverable error in mad_header_decode (%s)", mad_stream_errorstr(&_stream));
-				continue;
-			} else {
-				warning("MP3Stream: Unrecoverable error in mad_header_decode (%s)", mad_stream_errorstr(&_stream));
-				break;
-			}
-		}
-
-		// Sum up the total playback time so far
-		mad_timer_add(&_totalTime, _frame.header.duration);
-		break;
-	}
-
-	if (_stream.error != MAD_ERROR_NONE)
-		_state = MP3_STATE_EOS;
-}
-
-void MP3Stream::deinitStream() {
-	if (_state == MP3_STATE_INIT)
-		return;
-
-	// Deinit MAD
-	mad_synth_finish(&_synth);
-	mad_frame_finish(&_frame);
-	mad_stream_finish(&_stream);
-
-	_state = MP3_STATE_EOS;
-}
-
-static inline int scale_sample(mad_fixed_t sample) {
-	// round
-	sample += (1L << (MAD_F_FRACBITS - 16));
-
-	// clip
-	if (sample > MAD_F_ONE - 1)
-		sample = MAD_F_ONE - 1;
-	else if (sample < -MAD_F_ONE)
-		sample = -MAD_F_ONE;
-
-	// quantize and scale to not saturate when mixing a lot of channels
-	return sample >> (MAD_F_FRACBITS + 1 - 16);
-}
-
-int MP3Stream::readBuffer(int16 *buffer, const int numSamples) {
-	int samples = 0;
-	// Keep going as long as we have input available
-	while (samples < numSamples && _state != MP3_STATE_EOS) {
-		const int len = MIN(numSamples, samples + (int)(_synth.pcm.length - _posInFrame) * MAD_NCHANNELS(&_frame.header));
-		while (samples < len) {
-			*buffer++ = (int16)scale_sample(_synth.pcm.samples[0][_posInFrame]);
-			samples++;
-			if (MAD_NCHANNELS(&_frame.header) == 2) {
-				*buffer++ = (int16)scale_sample(_synth.pcm.samples[1][_posInFrame]);
-				samples++;
-			}
-			_posInFrame++;
-		}
-		if (_posInFrame >= _synth.pcm.length) {
-			// We used up all PCM data in the current frame -- read & decode more
-			decodeMP3Data();
-		}
-	}
-	return samples;
-}
-
-
-#pragma mark -
-#pragma mark --- MP3 factory functions ---
-#pragma mark -
-
-SeekableAudioStream *makeMP3Stream(
-	Common::SeekableReadStream *stream,
-	DisposeAfterUse::Flag disposeAfterUse) {
-
-#if defined(__PSP__)
-	SeekableAudioStream *s = 0;
-
-	if (Mp3PspStream::isOkToCreateStream())
-		s = new Mp3PspStream(stream, disposeAfterUse);
-
-	if (!s)	// go to regular MAD mp3 stream if ME fails
-		s = new MP3Stream(stream, disposeAfterUse);
-#else
-	SeekableAudioStream *s = new MP3Stream(stream, disposeAfterUse);
-#endif
-	if (s && s->endOfData()) {
-		delete s;
-		return 0;
-	} else {
-		return s;
-	}
-}
-
-} // End of namespace Audio
-
-#endif // #ifdef USE_MAD
diff --git a/sound/decoders/mp3.h b/sound/decoders/mp3.h
deleted file mode 100644
index 72bc6e1b3e..0000000000
--- a/sound/decoders/mp3.h
+++ /dev/null
@@ -1,76 +0,0 @@
-/* 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
- *  - draci
- *  - kyra
- *  - m4
- *  - mohawk
- *  - queen
- *  - saga
- *  - sci
- *  - scumm
- *  - sword1
- *  - sword2
- *  - touche
- *  - tucker
- */
-
-#ifndef SOUND_MP3_H
-#define SOUND_MP3_H
-
-#include "common/scummsys.h"
-#include "common/types.h"
-
-#ifdef USE_MAD
-
-namespace Common {
-	class SeekableReadStream;
-}
-
-namespace Audio {
-
-class AudioStream;
-class SeekableAudioStream;
-
-/**
- * Create a new SeekableAudioStream from the MP3 data in the given stream.
- * Allows for seeking (which is why we require a SeekableReadStream).
- *
- * @param stream			the SeekableReadStream from which to read the MP3 data
- * @param disposeAfterUse	whether to delete the stream after use
- * @return	a new SeekableAudioStream, or NULL, if an error occurred
- */
-SeekableAudioStream *makeMP3Stream(
-	Common::SeekableReadStream *stream,
-	DisposeAfterUse::Flag disposeAfterUse);
-
-} // End of namespace Audio
-
-#endif // #ifdef USE_MAD
-#endif // #ifndef SOUND_MP3_H
diff --git a/sound/decoders/raw.cpp b/sound/decoders/raw.cpp
deleted file mode 100644
index aedddbf6c4..0000000000
--- a/sound/decoders/raw.cpp
+++ /dev/null
@@ -1,356 +0,0 @@
-/* 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 "common/memstream.h"
-
-#include "sound/audiostream.h"
-#include "sound/mixer.h"
-#include "sound/decoders/raw.h"
-
-namespace Audio {
-
-// This used to be an inline template function, but
-// buggy template function handling in MSVC6 forced
-// us to go with the macro approach. So far this is
-// the only template function that MSVC6 seemed to
-// compile incorrectly. Knock on wood.
-#define READ_ENDIAN_SAMPLE(is16Bit, isUnsigned, ptr, isLE) \
-	((is16Bit ? (isLE ? READ_LE_UINT16(ptr) : READ_BE_UINT16(ptr)) : (*ptr << 8)) ^ (isUnsigned ? 0x8000 : 0))
-
-
-#pragma mark -
-#pragma mark --- RawStream ---
-#pragma mark -
-
-/**
- * This is a stream, which allows for playing raw PCM data from a stream.
- * It also features playback of multiple blocks from a given stream.
- */
-template<bool is16Bit, bool isUnsigned, bool isLE>
-class RawStream : public SeekableAudioStream {
-public:
-	RawStream(int rate, bool stereo, DisposeAfterUse::Flag disposeStream, Common::SeekableReadStream *stream, const RawStreamBlockList &blocks)
-		: _rate(rate), _isStereo(stereo), _playtime(0, rate), _stream(stream), _disposeAfterUse(disposeStream), _blocks(blocks), _curBlock(_blocks.begin()), _blockLeft(0), _buffer(0) {
-
-		assert(_blocks.size() > 0);
-
-		// Setup our buffer for readBuffer
-		_buffer = new byte[kSampleBufferLength * (is16Bit ? 2 : 1)];
-		assert(_buffer);
-
-		// Set current buffer state, playing first block
-		_stream->seek(_curBlock->pos, SEEK_SET);
-
-		// In case of an error we will stop (or rather
-		// not start) stream playback.
-		if (_stream->err()) {
-			_blockLeft = 0;
-			_curBlock = _blocks.end();
-		} else {
-			_blockLeft = _curBlock->len;
-		}
-
-		// Add up length of all blocks in order to caluclate total play time
-		int32 len = 0;
-		for (RawStreamBlockList::const_iterator i = _blocks.begin(); i != _blocks.end(); ++i) {
-			assert(i->len % (_isStereo ? 2 : 1) == 0);
-			len += i->len;
-		}
-
-		_playtime = Timestamp(0, len / (_isStereo ? 2 : 1), rate);
-	}
-
-	~RawStream() {
-		if (_disposeAfterUse == DisposeAfterUse::YES)
-			delete _stream;
-
-		delete[] _buffer;
-	}
-
-	int readBuffer(int16 *buffer, const int numSamples);
-
-	bool isStereo() const           { return _isStereo; }
-	bool endOfData() const          { return (_curBlock == _blocks.end()) && (_blockLeft == 0); }
-
-	int getRate() const         { return _rate; }
-	Timestamp getLength() const { return _playtime; }
-
-	bool seek(const Timestamp &where);
-private:
-	const int _rate;                               ///< Sample rate of stream
-	const bool _isStereo;                          ///< Whether this is an stereo stream
-	Timestamp _playtime;                           ///< Calculated total play time
-	Common::SeekableReadStream *_stream;           ///< Stream to read data from
-	const DisposeAfterUse::Flag _disposeAfterUse;  ///< Indicates whether the stream object should be deleted when this RawStream is destructed
-	const RawStreamBlockList _blocks;              ///< Audio block list
-
-	RawStreamBlockList::const_iterator _curBlock;  ///< Current audio block number
-	int32 _blockLeft;                              ///< How many bytes are still left in the current block
-
-	/**
-	 * Advance one block in the stream in case
-	 * the current one is empty.
-	 */
-	void updateBlockIfNeeded();
-
-	byte *_buffer;                                 ///< Buffer used in readBuffer
-	enum {
-		/**
-		 * How many samples we can buffer at once.
-		 *
-		 * TODO: Check whether this size suffices
-		 * for systems with slow disk I/O.
-		 */
-		kSampleBufferLength = 2048
-	};
-
-	/**
-	 * Fill the temporary sample buffer used in readBuffer.
-	 *
-	 * @param maxSamples Maximum samples to read.
-	 * @return actual count of samples read.
-	 */
-	int fillBuffer(int maxSamples);
-};
-
-template<bool is16Bit, bool isUnsigned, bool isLE>
-int RawStream<is16Bit, isUnsigned, isLE>::readBuffer(int16 *buffer, const int numSamples) {
-	int samplesLeft = numSamples;
-
-	while (samplesLeft > 0) {
-		// Try to read up to "samplesLeft" samples.
-		int len = fillBuffer(samplesLeft);
-
-		// In case we were not able to read any samples
-		// we will stop reading here.
-		if (!len)
-			break;
-
-		// Adjust the samples left to read.
-		samplesLeft -= len;
-
-		// Copy the data to the caller's buffer.
-		const byte *src = _buffer;
-		while (len-- > 0) {
-			*buffer++ = READ_ENDIAN_SAMPLE(is16Bit, isUnsigned, src, isLE);
-			src += (is16Bit ? 2 : 1);
-		}
-	}
-
-	return numSamples - samplesLeft;
-}
-
-template<bool is16Bit, bool isUnsigned, bool isLE>
-int RawStream<is16Bit, isUnsigned, isLE>::fillBuffer(int maxSamples) {
-	int bufferedSamples = 0;
-	byte *dst = _buffer;
-
-	// We can only read up to "kSampleBufferLength" samples
-	// so we take this into consideration, when trying to
-	// read up to maxSamples.
-	maxSamples = MIN<int>(kSampleBufferLength, maxSamples);
-
-	// We will only read up to maxSamples
-	while (maxSamples > 0 && !endOfData()) {
-		// Calculate how many samples we can safely read
-		// from the current block.
-		const int len = MIN<int>(maxSamples, _blockLeft);
-
-		// Try to read all the sample data and update the
-		// destination pointer.
-		const int bytesRead = _stream->read(dst, len * (is16Bit ? 2 : 1));
-		dst += bytesRead;
-
-		// Calculate how many samples we actually read.
-		const int samplesRead = bytesRead / (is16Bit ? 2 : 1);
-
-		// Update all status variables
-		bufferedSamples += samplesRead;
-		maxSamples -= samplesRead;
-		_blockLeft -= samplesRead;
-
-		// In case of an error we will stop
-		// stream playback.
-		if (_stream->err()) {
-			_blockLeft = 0;
-			_curBlock = _blocks.end();
-		}
-
-		// Advance to the next block in case the current
-		// one is already finished.
-		updateBlockIfNeeded();
-	}
-
-	return bufferedSamples;
-}
-
-template<bool is16Bit, bool isUnsigned, bool isLE>
-void RawStream<is16Bit, isUnsigned, isLE>::updateBlockIfNeeded() {
-	// Have we now finished this block? If so, read the next block
-	if (_blockLeft == 0 && _curBlock != _blocks.end()) {
-		// Next block
-		++_curBlock;
-
-		// Check whether we reached the end of the stream
-		// yet. In case we did not do this, we will just
-		// setup the next block as new block.
-		if (_curBlock != _blocks.end()) {
-			_stream->seek(_curBlock->pos, SEEK_SET);
-
-			// In case of an error we will stop
-			// stream playback.
-			if (_stream->err()) {
-				_blockLeft = 0;
-				_curBlock = _blocks.end();
-			} else {
-				_blockLeft = _curBlock->len;
-			}
-		}
-	}
-}
-
-template<bool is16Bit, bool isUnsigned, bool isLE>
-bool RawStream<is16Bit, isUnsigned, isLE>::seek(const Timestamp &where) {
-	_blockLeft = 0;
-	_curBlock = _blocks.end();
-
-	if (where > _playtime)
-		return false;
-
-	const uint32 seekSample = convertTimeToStreamPos(where, getRate(), isStereo()).totalNumberOfFrames();
-	uint32 curSample = 0;
-
-	// Search for the disk block in which the specific sample is placed
-	for (_curBlock = _blocks.begin(); _curBlock != _blocks.end(); ++_curBlock) {
-		uint32 nextBlockSample = curSample + _curBlock->len;
-
-		if (nextBlockSample > seekSample)
-			break;
-
-		curSample = nextBlockSample;
-	}
-
-	if (_curBlock == _blocks.end()) {
-		return ((seekSample - curSample) == 0);
-	} else {
-		const uint32 offset = seekSample - curSample;
-
-		_stream->seek(_curBlock->pos + offset * (is16Bit ? 2 : 1), SEEK_SET);
-
-		// In case of an error we will stop
-		// stream playback.
-		if (_stream->err()) {
-			_blockLeft = 0;
-			_curBlock = _blocks.end();
-		} else {
-			_blockLeft = _curBlock->len - offset;
-		}
-
-		return true;
-	}
-}
-
-#pragma mark -
-#pragma mark --- Raw stream factories ---
-#pragma mark -
-
-/* In the following, we use preprocessor / macro tricks to simplify the code
- * which instantiates the input streams. We used to use template functions for
- * this, but MSVC6 / EVC 3-4 (used for WinCE builds) are extremely buggy when it
- * comes to this feature of C++... so as a compromise we use macros to cut down
- * on the (source) code duplication a bit.
- * So while normally macro tricks are said to make maintenance harder, in this
- * particular case it should actually help it :-)
- */
-
-#define MAKE_RAW_STREAM(UNSIGNED) \
-		if (is16Bit) { \
-			if (isLE) \
-				return new RawStream<true, UNSIGNED, true>(rate, isStereo, disposeAfterUse, stream, blockList); \
-			else  \
-				return new RawStream<true, UNSIGNED, false>(rate, isStereo, disposeAfterUse, stream, blockList); \
-		} else \
-			return new RawStream<false, UNSIGNED, false>(rate, isStereo, disposeAfterUse, stream, blockList)
-
-SeekableAudioStream *makeRawStream(Common::SeekableReadStream *stream,
-                                   const RawStreamBlockList &blockList,
-                                   int rate,
-                                   byte flags,
-                                   DisposeAfterUse::Flag disposeAfterUse) {
-	const bool isStereo   = (flags & Audio::FLAG_STEREO) != 0;
-	const bool is16Bit    = (flags & Audio::FLAG_16BITS) != 0;
-	const bool isUnsigned = (flags & Audio::FLAG_UNSIGNED) != 0;
-	const bool isLE       = (flags & Audio::FLAG_LITTLE_ENDIAN) != 0;
-
-	if (blockList.empty()) {
-		warning("Empty block list passed to makeRawStream");
-		if (disposeAfterUse == DisposeAfterUse::YES)
-			delete stream;
-		return 0;
-	}
-
-	if (isUnsigned) {
-		MAKE_RAW_STREAM(true);
-	} else {
-		MAKE_RAW_STREAM(false);
-	}
-}
-
-SeekableAudioStream *makeRawStream(Common::SeekableReadStream *stream,
-                                   int rate, byte flags,
-                                   DisposeAfterUse::Flag disposeAfterUse) {
-	RawStreamBlockList blocks;
-	RawStreamBlock block;
-	block.pos = 0;
-
-	const bool isStereo   = (flags & Audio::FLAG_STEREO) != 0;
-	const bool is16Bit    = (flags & Audio::FLAG_16BITS) != 0;
-
-	assert(stream->size() % ((is16Bit ? 2 : 1) * (isStereo ? 2 : 1)) == 0);
-
-	block.len = stream->size() / (is16Bit ? 2 : 1);
-	blocks.push_back(block);
-
-	return makeRawStream(stream, blocks, rate, flags, disposeAfterUse);
-}
-
-SeekableAudioStream *makeRawStream(const byte *buffer, uint32 size,
-                                   int rate, byte flags,
-                                   DisposeAfterUse::Flag disposeAfterUse) {
-	return makeRawStream(new Common::MemoryReadStream(buffer, size, disposeAfterUse), rate, flags, DisposeAfterUse::YES);
-}
-
-SeekableAudioStream *makeRawDiskStream_OLD(Common::SeekableReadStream *stream, RawStreamBlock *block, int numBlocks,
-                                           int rate, byte flags, DisposeAfterUse::Flag disposeStream) {
-	assert(numBlocks > 0);
-	RawStreamBlockList blocks;
-	for (int i = 0; i < numBlocks; ++i)
-		blocks.push_back(block[i]);
-
-	return makeRawStream(stream, blocks, rate, flags, disposeStream);
-}
-
-} // End of namespace Audio
diff --git a/sound/decoders/raw.h b/sound/decoders/raw.h
deleted file mode 100644
index 3e9426012c..0000000000
--- a/sound/decoders/raw.h
+++ /dev/null
@@ -1,153 +0,0 @@
-/* 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_RAW_H
-#define SOUND_RAW_H
-
-#include "common/scummsys.h"
-#include "common/types.h"
-
-#include "common/list.h"
-
-
-namespace Common { class SeekableReadStream; }
-
-
-namespace Audio {
-
-class AudioStream;
-class SeekableAudioStream;
-
-/**
- * Various flags which can be bit-ORed and then passed to
- * makeRawStream and some other AudioStream factories
- * to control their behavior.
- *
- * Engine authors are advised not to rely on a certain value or
- * order of these flags (in particular, do not store them verbatim
- * in savestates).
- */
-enum RawFlags {
-	/** unsigned samples (default: signed) */
-	FLAG_UNSIGNED = 1 << 0,
-
-	/** sound is 16 bits wide (default: 8bit) */
-	FLAG_16BITS = 1 << 1,
-
-	/** samples are little endian (default: big endian) */
-	FLAG_LITTLE_ENDIAN = 1 << 2,
-
-	/** sound is in stereo (default: mono) */
-	FLAG_STEREO = 1 << 3
-};
-
-
-/**
- * Struct used to define the audio data to be played by a RawStream.
- */
-struct RawStreamBlock {
-	int32 pos;   ///< Position in stream of the block (in bytes of course!)
-	int32 len;   ///< Length of the block (in raw samples, not sample pairs!)
-};
-
-/**
- * List containing all blocks of a raw stream.
- * @see RawStreamBlock
- */
-typedef Common::List<RawStreamBlock> RawStreamBlockList;
-
-/**
- * Creates an audio stream, which plays from the given buffer.
- *
- * @param buffer Buffer to play from.
- * @param size   Size of the buffer in bytes.
- * @param rate   Rate of the sound data.
- * @param flags  Audio flags combination.
- * @see RawFlags
- * @param disposeAfterUse Whether to free the buffer after use (with free!).
- * @return The new SeekableAudioStream (or 0 on failure).
- */
-SeekableAudioStream *makeRawStream(const byte *buffer, uint32 size,
-                                   int rate, byte flags,
-                                   DisposeAfterUse::Flag disposeAfterUse = DisposeAfterUse::YES);
-
-/**
- * Creates an audio stream, which plays from the given stream.
- *
- * @param stream Stream object to play from.
- * @param rate   Rate of the sound data.
- * @param flags  Audio flags combination.
- * @see RawFlags
- * @param disposeAfterUse Whether to delete the stream after use.
- * @return The new SeekableAudioStream (or 0 on failure).
- */
-SeekableAudioStream *makeRawStream(Common::SeekableReadStream *stream,
-                                   int rate, byte flags,
-                                   DisposeAfterUse::Flag disposeAfterUse = DisposeAfterUse::YES);
-
-/**
- * Creates an audio stream, which plays from the given stream.
- *
- * @param stream Stream object to play from.
- * @param blockList List of blocks to play.
- * @see RawDiskStreamAudioBlock
- * @see RawStreamBlockList
- * @param rate Rate of the sound data.
- * @param flags Audio flags combination.
- * @see RawFlags
- * @param disposeAfterUse Whether to delete the stream after use.
- * @return The new SeekableAudioStream (or 0 on failure).
- */
-SeekableAudioStream *makeRawStream(Common::SeekableReadStream *stream,
-                                   const RawStreamBlockList &blockList,
-                                   int rate,
-                                   byte flags,
-                                   DisposeAfterUse::Flag disposeAfterUse = DisposeAfterUse::YES);
-
-/**
- * NOTE:
- * This API is considered deprecated.
- *
- * Creates a audio stream, which plays from given stream.
- *
- * @param stream Stream to play from
- * @param block Pointer to an RawStreamBlock array
- * @see RawStreamBlock
- * @param numBlocks Number of blocks.
- * @param rate The rate
- * @param flags Flags combination.
- * @see RawFlags
- * @param disposeStream Whether the "stream" object should be destroyed after playback.
- * @return The new SeekableAudioStream (or 0 on failure).
- */
-SeekableAudioStream *makeRawDiskStream_OLD(Common::SeekableReadStream *stream,
-		RawStreamBlock *block, int numBlocks,
-		int rate, byte flags,
-		DisposeAfterUse::Flag disposeStream);
-
-
-} // End of namespace Audio
-
-#endif
diff --git a/sound/decoders/vag.cpp b/sound/decoders/vag.cpp
deleted file mode 100644
index d3f0811f2b..0000000000
--- a/sound/decoders/vag.cpp
+++ /dev/null
@@ -1,150 +0,0 @@
-/* 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 "sound/decoders/vag.h"
-#include "sound/audiostream.h"
-#include "common/stream.h"
-
-namespace Audio {
-
-class VagStream : public Audio::RewindableAudioStream {
-public:
-	VagStream(Common::SeekableReadStream *stream, int rate);
-	~VagStream();
-
-	bool isStereo() const { return false; }
-	bool endOfData() const { return _stream->pos() == _stream->size(); }
-	int getRate() const { return _rate; }
-	int readBuffer(int16 *buffer, const int numSamples);
-
-	bool rewind();
-private:
-	Common::SeekableReadStream *_stream;
-
-	byte _predictor;
-	double _samples[28];
-	byte _samplesRemaining;
-	int _rate;
-	double _s1, _s2;
-};
-
-VagStream::VagStream(Common::SeekableReadStream *stream, int rate) : _stream(stream) {
-	_samplesRemaining = 0;
-	_predictor = 0;
-	_s1 = _s2 = 0.0;
-	_rate = rate;
-}
-
-
-VagStream::~VagStream() {
-	delete _stream;
-}
-
-static const double s_vagDataTable[5][2] =
-	{
-		{  0.0, 0.0 },
-		{  60.0 / 64.0,  0.0 },
-		{  115.0 / 64.0, -52.0 / 64.0 },
-		{  98.0 / 64.0, -55.0 / 64.0 },
-		{  122.0 / 64.0, -60.0 / 64.0 }
-	};
-
-int VagStream::readBuffer(int16 *buffer, const int numSamples) {
-	int32 samplesDecoded = 0;
-
-	if (_samplesRemaining) {
-		byte i = 0;
-
-		for (i = 28 - _samplesRemaining; i < 28 && samplesDecoded < numSamples; i++) {
-			_samples[i] = _samples[i] + _s1 * s_vagDataTable[_predictor][0] + _s2 * s_vagDataTable[_predictor][1];
-			_s2 = _s1;
-			_s1 = _samples[i];
-			int16 d = (int) (_samples[i] + 0.5);
-			buffer[samplesDecoded] = d;
-			samplesDecoded++;
-		}
-
-#if 0
-		assert(i == 28); // We're screwed if this fails :P
-#endif
-		// This might mean the file is corrupted, or that the stream has
-		// been closed.
-		if (i != 28) return 0;
-
-		_samplesRemaining = 0;
-	}
-
-	while (samplesDecoded < numSamples) {
-		byte i = 0;
-
-		_predictor = _stream->readByte();
-		byte shift = _predictor & 0xf;
-		_predictor >>= 4;
-
-		if (_stream->readByte() == 7)
-			return samplesDecoded;
-
-		for (i = 0; i < 28; i += 2) {
-			byte d = _stream->readByte();
-			int16 s = (d & 0xf) << 12;
-			if (s & 0x8000)
-				s |= 0xffff0000;
-			_samples[i] = (double)(s >> shift);
-			s = (d & 0xf0) << 8;
-			if (s & 0x8000)
-				s |= 0xffff0000;
-			_samples[i + 1] = (double)(s >> shift);
-		}
-
-		for (i = 0; i < 28 && samplesDecoded < numSamples; i++) {
-			_samples[i] = _samples[i] + _s1 * s_vagDataTable[_predictor][0] + _s2 * s_vagDataTable[_predictor][1];
-			_s2 = _s1;
-			_s1 = _samples[i];
-			int16 d = (int) (_samples[i] + 0.5);
-			buffer[samplesDecoded] = d;
-			samplesDecoded++;
-		}
-
-		if (i != 27)
-			_samplesRemaining = 28 - i;
-	}
-
-	return samplesDecoded;
-}
-
-bool VagStream::rewind() {
-	_stream->seek(0);
-	_samplesRemaining = 0;
-	_predictor = 0;
-	_s1 = _s2 = 0.0;
-
-	return true;
-}
-
-RewindableAudioStream *makeVagStream(Common::SeekableReadStream *stream, int rate) {
-	return new VagStream(stream, rate);
-}
-
-}
diff --git a/sound/decoders/vag.h b/sound/decoders/vag.h
deleted file mode 100644
index cdf91a8ea1..0000000000
--- a/sound/decoders/vag.h
+++ /dev/null
@@ -1,60 +0,0 @@
-/* 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:
- * - sword1 (PSX port of the game)
- * - sword2 (PSX port of the game)
- * - tinsel (PSX port of the game)
- */
-
-#ifndef SOUND_VAG_H
-#define SOUND_VAG_H
-
-namespace Common {
-	class SeekableReadStream;
-}
-
-namespace Audio {
-
-class AudioStream;
-class RewindableAudioStream;
-
-/**
- * Takes an input stream containing Vag sound data and creates
- * an RewindableAudioStream from that.
- *
- * @param stream            the SeekableReadStream from which to read the ADPCM data
- * @param rate              the sampling rate
- * @return   a new RewindableAudioStream, or NULL, if an error occurred
- */
-RewindableAudioStream *makeVagStream(
-	Common::SeekableReadStream *stream,
-	int rate = 11025);
-
-} // End of namespace Sword1
-
-#endif
diff --git a/sound/decoders/voc.cpp b/sound/decoders/voc.cpp
deleted file mode 100644
index e9af7ece3f..0000000000
--- a/sound/decoders/voc.cpp
+++ /dev/null
@@ -1,403 +0,0 @@
-/* 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 "common/util.h"
-#include "common/stream.h"
-
-#include "sound/audiostream.h"
-#include "sound/mixer.h"
-#include "sound/decoders/raw.h"
-#include "sound/decoders/voc.h"
-
-
-namespace Audio {
-
-int getSampleRateFromVOCRate(int vocSR) {
-	if (vocSR == 0xa5 || vocSR == 0xa6) {
-		return 11025;
-	} else if (vocSR == 0xd2 || vocSR == 0xd3) {
-		return 22050;
-	} else {
-		int sr = 1000000L / (256L - vocSR);
-		// inexact sampling rates occur e.g. in the kitchen in Monkey Island,
-		// very easy to reach right from the start of the game.
-		//warning("inexact sample rate used: %i (0x%x)", sr, vocSR);
-		return sr;
-	}
-}
-
-static byte *loadVOCFromStream(Common::ReadStream &stream, int &size, int &rate, int &loops, int &begin_loop, int &end_loop) {
-	VocFileHeader fileHeader;
-
-	debug(2, "loadVOCFromStream");
-
-	if (stream.read(&fileHeader, 8) != 8)
-		goto invalid;
-
-	if (!memcmp(&fileHeader, "VTLK", 4)) {
-		if (stream.read(&fileHeader, sizeof(VocFileHeader)) != sizeof(VocFileHeader))
-			goto invalid;
-	} else if (!memcmp(&fileHeader, "Creative", 8)) {
-		if (stream.read(((byte *)&fileHeader) + 8, sizeof(VocFileHeader) - 8) != sizeof(VocFileHeader) - 8)
-			goto invalid;
-	} else {
-	invalid:;
-		warning("loadVOCFromStream: Invalid header");
-		return NULL;
-	}
-
-	if (memcmp(fileHeader.desc, "Creative Voice File", 19) != 0)
-		error("loadVOCFromStream: Invalid header");
-	if (fileHeader.desc[19] != 0x1A)
-		debug(3, "loadVOCFromStream: Partially invalid header");
-
-	int32 offset = FROM_LE_16(fileHeader.datablock_offset);
-	int16 version = FROM_LE_16(fileHeader.version);
-	int16 code = FROM_LE_16(fileHeader.id);
-	assert(offset == sizeof(VocFileHeader));
-	// 0x100 is an invalid VOC version used by German version of DOTT (Disk) and
-	// French version of Simon the Sorcerer 2 (CD)
-	assert(version == 0x010A || version == 0x0114 || version == 0x0100);
-	assert(code == ~version + 0x1234);
-
-	int len;
-	byte *ret_sound = 0;
-	size = 0;
-	begin_loop = 0;
-	end_loop = 0;
-
-	while ((code = stream.readByte())) {
-		len = stream.readByte();
-		len |= stream.readByte() << 8;
-		len |= stream.readByte() << 16;
-
-		debug(2, "Block code %d, len %d", code, len);
-
-		switch (code) {
-		case 1:
-		case 9: {
-			int packing;
-			if (code == 1) {
-				int time_constant = stream.readByte();
-				packing = stream.readByte();
-				len -= 2;
-				rate = getSampleRateFromVOCRate(time_constant);
-			} else {
-				rate = stream.readUint32LE();
-				int bits = stream.readByte();
-				int channels = stream.readByte();
-				if (bits != 8 || channels != 1) {
-					warning("Unsupported VOC file format (%d bits per sample, %d channels)", bits, channels);
-					break;
-				}
-				packing = stream.readUint16LE();
-				stream.readUint32LE();
-				len -= 12;
-			}
-			debug(9, "VOC Data Block: %d, %d, %d", rate, packing, len);
-			if (packing == 0) {
-				if (size) {
-					ret_sound = (byte *)realloc(ret_sound, size + len);
-				} else {
-					ret_sound = (byte *)malloc(len);
-				}
-				stream.read(ret_sound + size, len);
-				size += len;
-				begin_loop = size;
-				end_loop = size;
-			} else {
-				warning("VOC file packing %d unsupported", packing);
-			}
-			} break;
-		case 3: // silence
-			// occur with a few Igor sounds, voc file starts with a silence block with a
-			// frequency different from the data block. Just ignore fow now (implementing
-			// it wouldn't make a big difference anyway...)
-			assert(len == 3);
-			stream.readUint16LE();
-			stream.readByte();
-			break;
-		case 6:	// begin of loop
-			assert(len == 2);
-			loops = stream.readUint16LE();
-			break;
-		case 7:	// end of loop
-			assert(len == 0);
-			break;
-		case 8: { // "Extended"
-			// This occures in the LoL Intro demo.
-			// This block overwrites the next parameters of a block 1 "Sound data".
-			// To assure we never get any bad data here, we will assert in case
-			// this tries to define a stereo sound block or tries to use something
-			// different than 8bit unsigned sound data.
-			// TODO: Actually we would need to check the frequency divisor (the
-			// first word) here too. It is used in the following equation:
-			// sampleRate = 256000000/(channels * (65536 - frequencyDivisor))
-			assert(len == 4);
-			stream.readUint16LE();
-			uint8 codec = stream.readByte();
-			uint8 channels = stream.readByte() + 1;
-			assert(codec == 0 && channels == 1);
-			} break;
-		default:
-			warning("Unhandled code %d in VOC file (len %d)", code, len);
-			return ret_sound;
-		}
-	}
-	debug(4, "VOC Data Size : %d", size);
-	return ret_sound;
-}
-
-byte *loadVOCFromStream(Common::ReadStream &stream, int &size, int &rate) {
-	int loops, begin_loop, end_loop;
-	return loadVOCFromStream(stream, size, rate, loops, begin_loop, end_loop);
-}
-
-
-#ifdef STREAM_AUDIO_FROM_DISK
-
-int parseVOCFormat(Common::SeekableReadStream& stream, RawStreamBlock* block, int &rate, int &loops, int &begin_loop, int &end_loop) {
-	VocFileHeader fileHeader;
-	int currentBlock = 0;
-	int size = 0;
-
-	debug(2, "parseVOCFormat");
-
-	if (stream.read(&fileHeader, 8) != 8)
-		goto invalid;
-
-	if (!memcmp(&fileHeader, "VTLK", 4)) {
-		if (stream.read(&fileHeader, sizeof(VocFileHeader)) != sizeof(VocFileHeader))
-			goto invalid;
-	} else if (!memcmp(&fileHeader, "Creative", 8)) {
-		if (stream.read(((byte *)&fileHeader) + 8, sizeof(VocFileHeader) - 8) != sizeof(VocFileHeader) - 8)
-			goto invalid;
-	} else {
-	invalid:;
-		warning("loadVOCFromStream: Invalid header");
-		return 0;
-	}
-
-	if (memcmp(fileHeader.desc, "Creative Voice File", 19) != 0)
-		error("loadVOCFromStream: Invalid header");
-	if (fileHeader.desc[19] != 0x1A)
-		debug(3, "loadVOCFromStream: Partially invalid header");
-
-	int32 offset = FROM_LE_16(fileHeader.datablock_offset);
-	int16 version = FROM_LE_16(fileHeader.version);
-	int16 code = FROM_LE_16(fileHeader.id);
-	assert(offset == sizeof(VocFileHeader));
-	// 0x100 is an invalid VOC version used by German version of DOTT (Disk) and
-	// French version of Simon the Sorcerer 2 (CD)
-	assert(version == 0x010A || version == 0x0114 || version == 0x0100);
-	assert(code == ~version + 0x1234);
-
-	int len;
-	size = 0;
-	begin_loop = 0;
-	end_loop = 0;
-
-	while ((code = stream.readByte())) {
-		len = stream.readByte();
-		len |= stream.readByte() << 8;
-		len |= stream.readByte() << 16;
-
-		debug(2, "Block code %d, len %d", code, len);
-
-		switch (code) {
-		case 1:
-		case 9: {
-			int packing;
-			if (code == 1) {
-				int time_constant = stream.readByte();
-				packing = stream.readByte();
-				len -= 2;
-				rate = getSampleRateFromVOCRate(time_constant);
-			} else {
-				rate = stream.readUint32LE();
-				int bits = stream.readByte();
-				int channels = stream.readByte();
-				if (bits != 8 || channels != 1) {
-					warning("Unsupported VOC file format (%d bits per sample, %d channels)", bits, channels);
-					break;
-				}
-				packing = stream.readUint16LE();
-				stream.readUint32LE();
-				len -= 12;
-			}
-			debug(9, "VOC Data Block: %d, %d, %d", rate, packing, len);
-			if (packing == 0) {
-
-				// Found a data block - so add it to the block list
-				block[currentBlock].pos = stream.pos();
-				block[currentBlock].len = len;
-				currentBlock++;
-
-				stream.seek(len, SEEK_CUR);
-
-				size += len;
-				begin_loop = size;
-				end_loop = size;
-			} else {
-				warning("VOC file packing %d unsupported", packing);
-			}
-			} break;
-		case 3: // silence
-			// occur with a few Igor sounds, voc file starts with a silence block with a
-			// frequency different from the data block. Just ignore fow now (implementing
-			// it wouldn't make a big difference anyway...)
-			assert(len == 3);
-			stream.readUint16LE();
-			stream.readByte();
-			break;
-		case 6:	// begin of loop
-			assert(len == 2);
-			loops = stream.readUint16LE();
-			break;
-		case 7:	// end of loop
-			assert(len == 0);
-			break;
-		case 8: // "Extended"
-			// This occures in the LoL Intro demo. This block can usually be used to create stereo
-			// sound, but the LoL intro has only an empty block, thus this dummy implementation will
-			// work.
-			assert(len == 4);
-			stream.readUint16LE();
-			stream.readByte();
-			stream.readByte();
-			break;
-		default:
-			warning("Unhandled code %d in VOC file (len %d)", code, len);
-			return 0;
-		}
-	}
-	debug(4, "VOC Data Size : %d", size);
-	return currentBlock;
-}
-
-AudioStream *makeVOCDiskStream(Common::SeekableReadStream *stream, byte flags, DisposeAfterUse::Flag disposeAfterUse) {
-	const int MAX_AUDIO_BLOCKS = 256;
-
-	RawStreamBlock *block = new RawStreamBlock[MAX_AUDIO_BLOCKS];
-	int rate, loops, begin_loop, end_loop;
-
-	int numBlocks = parseVOCFormat(*stream, block, rate, loops, begin_loop, end_loop);
-
-	AudioStream *audioStream = 0;
-
-	// Create an audiostream from the data. Note the numBlocks may be 0,
-	// e.g. when invalid data is encountered. See bug #2890038.
-	if (numBlocks)
-		audioStream = makeRawDiskStream_OLD(stream, block, numBlocks, rate, flags, disposeAfterUse/*, begin_loop, end_loop*/);
-
-	delete[] block;
-
-	return audioStream;
-}
-
-SeekableAudioStream *makeVOCDiskStreamNoLoop(Common::SeekableReadStream *stream, byte flags, DisposeAfterUse::Flag disposeAfterUse) {
-	const int MAX_AUDIO_BLOCKS = 256;
-
-	RawStreamBlock *block = new RawStreamBlock[MAX_AUDIO_BLOCKS];
-	int rate, loops, begin_loop, end_loop;
-
-	int numBlocks = parseVOCFormat(*stream, block, rate, loops, begin_loop, end_loop);
-
-	SeekableAudioStream *audioStream = 0;
-
-	// Create an audiostream from the data. Note the numBlocks may be 0,
-	// e.g. when invalid data is encountered. See bug #2890038.
-	if (numBlocks)
-		audioStream = makeRawDiskStream_OLD(stream, block, numBlocks, rate, flags, disposeAfterUse);
-
-	delete[] block;
-
-	return audioStream;
-}
-
-#endif
-
-
-AudioStream *makeVOCStream(Common::SeekableReadStream *stream, byte flags, uint loopStart, uint loopEnd, DisposeAfterUse::Flag disposeAfterUse) {
-#ifdef STREAM_AUDIO_FROM_DISK
-	return makeVOCDiskStream(stream, flags, disposeAfterUse);
-#else
-	int size, rate;
-
-	byte *data = loadVOCFromStream(*stream, size, rate);
-
-	if (!data) {
-		if (disposeAfterUse == DisposeAfterUse::YES)
-			delete stream;
-		return 0;
-	}
-
-	SeekableAudioStream *s = Audio::makeRawStream(data, size, rate, flags);
-
-	if (loopStart != loopEnd) {
-		const bool isStereo   = (flags & Audio::FLAG_STEREO) != 0;
-		const bool is16Bit    = (flags & Audio::FLAG_16BITS) != 0;
-
-		if (loopEnd == 0)
-			loopEnd = size;
-		assert(loopStart <= loopEnd);
-		assert(loopEnd <= (uint)size);
-
-		// Verify the buffer sizes are sane
-		if (is16Bit && isStereo)
-			assert((loopStart & 3) == 0 && (loopEnd & 3) == 0);
-		else if (is16Bit || isStereo)
-			assert((loopStart & 1) == 0 && (loopEnd & 1) == 0);
-
-		const uint32 extRate = s->getRate() * (is16Bit ? 2 : 1) * (isStereo ? 2 : 1);
-
-		return new SubLoopingAudioStream(s, 0, Timestamp(0, loopStart, extRate), Timestamp(0, loopEnd, extRate));
-	} else {
-		return s;
-	}
-#endif
-}
-
-SeekableAudioStream *makeVOCStream(Common::SeekableReadStream *stream, byte flags, DisposeAfterUse::Flag disposeAfterUse) {
-#ifdef STREAM_AUDIO_FROM_DISK
-	return makeVOCDiskStreamNoLoop(stream, flags, disposeAfterUse);
-#else
-	int size, rate;
-
-	byte *data = loadVOCFromStream(*stream, size, rate);
-
-	if (!data) {
-		if (disposeAfterUse == DisposeAfterUse::YES)
-			delete stream;
-		return 0;
-	}
-
-	return makeRawStream(data, size, rate, flags);
-#endif
-}
-
-} // End of namespace Audio
diff --git a/sound/decoders/voc.h b/sound/decoders/voc.h
deleted file mode 100644
index 82cc261f2c..0000000000
--- a/sound/decoders/voc.h
+++ /dev/null
@@ -1,107 +0,0 @@
-/* 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
- *  - drascula
- *  - kyra
- *  - made
- *  - saga
- *  - scumm
- *  - touche
- */
-
-#ifndef SOUND_VOC_H
-#define SOUND_VOC_H
-
-#include "common/scummsys.h"
-#include "common/types.h"
-
-namespace Common { class ReadStream; }
-namespace Common { class SeekableReadStream; }
-
-namespace Audio {
-
-class AudioStream;
-class SeekableAudioStream;
-
-
-#include "common/pack-start.h"	// START STRUCT PACKING
-
-struct VocFileHeader {
-	uint8 desc[20];
-	uint16 datablock_offset;
-	uint16 version;
-	uint16 id;
-} PACKED_STRUCT;
-
-struct VocBlockHeader {
-	uint8 blocktype;
-	uint8 size[3];
-	uint8 sr;
-	uint8 pack;
-} PACKED_STRUCT;
-
-#include "common/pack-end.h"	// END STRUCT PACKING
-
-/**
- * Take a sample rate parameter as it occurs in a VOC sound header, and
- * return the corresponding sample frequency.
- *
- * This method has special cases for the standard rates of 11025 and 22050 kHz,
- * which due to limitations of the format, cannot be encoded exactly in a VOC
- * file. As a consequence, many game files have sound data sampled with those
- * rates, but the VOC marks them incorrectly as 11111 or 22222 kHz. This code
- * works around that and "unrounds" the sampling rates.
- */
-extern int getSampleRateFromVOCRate(int vocSR);
-
-/**
- * Try to load a VOC from the given stream. Returns a pointer to memory
- * containing the PCM sample data (allocated with malloc). It is the callers
- * responsibility to dellocate that data again later on! Currently this
- * function only supports uncompressed raw PCM data.
- */
-extern byte *loadVOCFromStream(Common::ReadStream &stream, int &size, int &rate);
-
-/**
- * Try to load a VOC from the given seekable stream and create an AudioStream
- * from that data. Currently this function only supports uncompressed raw PCM
- * data. Optionally supports (infinite) looping of a portion of the data.
- *
- * This function uses loadVOCFromStream() internally.
- */
-AudioStream *makeVOCStream(Common::SeekableReadStream *stream, byte flags = 0, uint loopStart = 0, uint loopEnd = 0, DisposeAfterUse::Flag disposeAfterUse = DisposeAfterUse::NO);
-
-/**
- * This does not use any of the looping features of VOC files!
- */
-SeekableAudioStream *makeVOCStream(Common::SeekableReadStream *stream, byte flags, DisposeAfterUse::Flag disposeAfterUse);
-
-} // End of namespace Audio
-
-#endif
diff --git a/sound/decoders/vorbis.cpp b/sound/decoders/vorbis.cpp
deleted file mode 100644
index 425eb6b751..0000000000
--- a/sound/decoders/vorbis.cpp
+++ /dev/null
@@ -1,262 +0,0 @@
-/* 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.
- *
- * $URL$
- * $Id$
- *
- */
-
-// Disable symbol overrides for FILE and fseek as those are used in the
-// Vorbis headers.
-#define FORBIDDEN_SYMBOL_EXCEPTION_FILE
-#define FORBIDDEN_SYMBOL_EXCEPTION_fseek
-
-#include "sound/decoders/vorbis.h"
-
-#ifdef USE_VORBIS
-
-#include "common/debug.h"
-#include "common/stream.h"
-#include "common/util.h"
-
-#include "sound/audiostream.h"
-
-#ifdef USE_TREMOR
-#if defined(__GP32__) // custom libtremor locations
-#include <ivorbisfile.h>
-#else
-#include <tremor/ivorbisfile.h>
-#endif
-#else
-#include <vorbis/vorbisfile.h>
-#endif
-
-
-namespace Audio {
-
-// These are wrapper functions to allow using a SeekableReadStream object to
-// provide data to the OggVorbis_File object.
-
-static size_t read_stream_wrap(void *ptr, size_t size, size_t nmemb, void *datasource) {
-	Common::SeekableReadStream *stream = (Common::SeekableReadStream *)datasource;
-
-	uint32 result = stream->read(ptr, size * nmemb);
-
-	return result / size;
-}
-
-static int seek_stream_wrap(void *datasource, ogg_int64_t offset, int whence) {
-	Common::SeekableReadStream *stream = (Common::SeekableReadStream *)datasource;
-	stream->seek((int32)offset, whence);
-	return stream->pos();
-}
-
-static int close_stream_wrap(void *datasource) {
-	// Do nothing -- we leave it up to the VorbisStream to free memory as appropriate.
-	return 0;
-}
-
-static long tell_stream_wrap(void *datasource) {
-	Common::SeekableReadStream *stream = (Common::SeekableReadStream *)datasource;
-	return stream->pos();
-}
-
-static ov_callbacks g_stream_wrap = {
-	read_stream_wrap, seek_stream_wrap, close_stream_wrap, tell_stream_wrap
-};
-
-
-
-#pragma mark -
-#pragma mark --- Ogg Vorbis stream ---
-#pragma mark -
-
-
-class VorbisStream : public SeekableAudioStream {
-protected:
-	Common::SeekableReadStream *_inStream;
-	DisposeAfterUse::Flag _disposeAfterUse;
-
-	bool _isStereo;
-	int _rate;
-
-	Timestamp _length;
-
-	OggVorbis_File _ovFile;
-
-	int16 _buffer[4096];
-	const int16 *_bufferEnd;
-	const int16 *_pos;
-
-public:
-	// startTime / duration are in milliseconds
-	VorbisStream(Common::SeekableReadStream *inStream, DisposeAfterUse::Flag dispose);
-	~VorbisStream();
-
-	int readBuffer(int16 *buffer, const int numSamples);
-
-	bool endOfData() const		{ return _pos >= _bufferEnd; }
-	bool isStereo() const		{ return _isStereo; }
-	int getRate() const			{ return _rate; }
-
-	bool seek(const Timestamp &where);
-	Timestamp getLength() const { return _length; }
-protected:
-	bool refill();
-};
-
-VorbisStream::VorbisStream(Common::SeekableReadStream *inStream, DisposeAfterUse::Flag dispose) :
-	_inStream(inStream),
-	_disposeAfterUse(dispose),
-	_length(0, 1000),
-	_bufferEnd(_buffer + ARRAYSIZE(_buffer)) {
-
-	int res = ov_open_callbacks(inStream, &_ovFile, NULL, 0, g_stream_wrap);
-	if (res < 0) {
-		warning("Could not create Vorbis stream (%d)", res);
-		_pos = _bufferEnd;
-		return;
-	}
-
-	// Read in initial data
-	if (!refill())
-		return;
-
-	// Setup some header information
-	_isStereo = ov_info(&_ovFile, -1)->channels >= 2;
-	_rate = ov_info(&_ovFile, -1)->rate;
-
-#ifdef USE_TREMOR
-	_length = Timestamp(ov_time_total(&_ovFile, -1), getRate());
-#else
-	_length = Timestamp(uint32(ov_time_total(&_ovFile, -1) * 1000.0), getRate());
-#endif
-}
-
-VorbisStream::~VorbisStream() {
-	ov_clear(&_ovFile);
-	if (_disposeAfterUse == DisposeAfterUse::YES)
-		delete _inStream;
-}
-
-int VorbisStream::readBuffer(int16 *buffer, const int numSamples) {
-	int samples = 0;
-	while (samples < numSamples && _pos < _bufferEnd) {
-		const int len = MIN(numSamples - samples, (int)(_bufferEnd - _pos));
-		memcpy(buffer, _pos, len * 2);
-		buffer += len;
-		_pos += len;
-		samples += len;
-		if (_pos >= _bufferEnd) {
-			if (!refill())
-				break;
-		}
-	}
-	return samples;
-}
-
-bool VorbisStream::seek(const Timestamp &where) {
-	// Vorbisfile uses the sample pair number, thus we always use "false" for the isStereo parameter
-	// of the convertTimeToStreamPos helper.
-	int res = ov_pcm_seek(&_ovFile, convertTimeToStreamPos(where, getRate(), false).totalNumberOfFrames());
-	if (res) {
-		warning("Error seeking in Vorbis stream (%d)", res);
-		_pos = _bufferEnd;
-		return false;
-	}
-
-	return refill();
-}
-
-bool VorbisStream::refill() {
-	// Read the samples
-	uint len_left = sizeof(_buffer);
-	char *read_pos = (char *)_buffer;
-
-	while (len_left > 0) {
-		long result;
-
-#ifdef USE_TREMOR
-		// Tremor ov_read() always returns data as signed 16 bit interleaved PCM
-		// in host byte order. As such, it does not take arguments to request
-		// specific signedness, byte order or bit depth as in Vorbisfile.
-		result = ov_read(&_ovFile, read_pos, len_left,
-						NULL);
-#else
-#ifdef SCUMM_BIG_ENDIAN
-		result = ov_read(&_ovFile, read_pos, len_left,
-						1,
-						2,	// 16 bit
-						1,	// signed
-						NULL);
-#else
-		result = ov_read(&_ovFile, read_pos, len_left,
-						0,
-						2,	// 16 bit
-						1,	// signed
-						NULL);
-#endif
-#endif
-		if (result == OV_HOLE) {
-			// Possibly recoverable, just warn about it
-			warning("Corrupted data in Vorbis file");
-		} else if (result == 0) {
-			//warning("End of file while reading from Vorbis file");
-			//_pos = _bufferEnd;
-			//return false;
-			break;
-		} else if (result < 0) {
-			warning("Error reading from Vorbis stream (%d)", int(result));
-			_pos = _bufferEnd;
-			// Don't delete it yet, that causes problems in
-			// the CD player emulation code.
-			return false;
-		} else {
-			len_left -= result;
-			read_pos += result;
-		}
-	}
-
-	_pos = _buffer;
-	_bufferEnd = (int16 *)read_pos;
-
-	return true;
-}
-
-
-#pragma mark -
-#pragma mark --- Ogg Vorbis factory functions ---
-#pragma mark -
-
-SeekableAudioStream *makeVorbisStream(
-	Common::SeekableReadStream *stream,
-	DisposeAfterUse::Flag disposeAfterUse) {
-	SeekableAudioStream *s = new VorbisStream(stream, disposeAfterUse);
-	if (s && s->endOfData()) {
-		delete s;
-		return 0;
-	} else {
-		return s;
-	}
-}
-
-} // End of namespace Audio
-
-#endif // #ifdef USE_VORBIS
diff --git a/sound/decoders/vorbis.h b/sound/decoders/vorbis.h
deleted file mode 100644
index 7cc395cccb..0000000000
--- a/sound/decoders/vorbis.h
+++ /dev/null
@@ -1,75 +0,0 @@
-/* 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
- *  - draci
- *  - kyra
- *  - m4
- *  - queen
- *  - saga
- *  - sci
- *  - scumm
- *  - sword1
- *  - sword2
- *  - touche
- *  - tucker
- */
-
-#ifndef SOUND_VORBIS_H
-#define SOUND_VORBIS_H
-
-#include "common/scummsys.h"
-#include "common/types.h"
-
-#ifdef USE_VORBIS
-
-namespace Common {
-	class SeekableReadStream;
-}
-
-namespace Audio {
-
-class AudioStream;
-class SeekableAudioStream;
-
-/**
- * Create a new SeekableAudioStream from the Ogg Vorbis data in the given stream.
- * Allows for seeking (which is why we require a SeekableReadStream).
- *
- * @param stream			the SeekableReadStream from which to read the Ogg Vorbis data
- * @param disposeAfterUse	whether to delete the stream after use
- * @return	a new SeekableAudioStream, or NULL, if an error occurred
- */
-SeekableAudioStream *makeVorbisStream(
-	Common::SeekableReadStream *stream,
-	DisposeAfterUse::Flag disposeAfterUse);
-
-} // End of namespace Audio
-
-#endif // #ifdef USE_VORBIS
-#endif // #ifndef SOUND_VORBIS_H
diff --git a/sound/decoders/wave.cpp b/sound/decoders/wave.cpp
deleted file mode 100644
index fcaace5301..0000000000
--- a/sound/decoders/wave.cpp
+++ /dev/null
@@ -1,194 +0,0 @@
-/* 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/util.h"
-#include "common/stream.h"
-
-#include "sound/audiostream.h"
-#include "sound/mixer.h"
-#include "sound/decoders/wave.h"
-#include "sound/decoders/adpcm.h"
-#include "sound/decoders/raw.h"
-
-namespace Audio {
-
-bool loadWAVFromStream(Common::SeekableReadStream &stream, int &size, int &rate, byte &flags, uint16 *wavType, int *blockAlign_) {
-	const int32 initialPos = stream.pos();
-	byte buf[4+1];
-
-	buf[4] = 0;
-
-	stream.read(buf, 4);
-	if (memcmp(buf, "RIFF", 4) != 0) {
-		warning("getWavInfo: No 'RIFF' header");
-		return false;
-	}
-
-	int32 wavLength = stream.readUint32LE();
-
-	stream.read(buf, 4);
-	if (memcmp(buf, "WAVE", 4) != 0) {
-		warning("getWavInfo: No 'WAVE' header");
-		return false;
-	}
-
-	stream.read(buf, 4);
-	if (memcmp(buf, "fmt ", 4) != 0) {
-		warning("getWavInfo: No 'fmt' header");
-		return false;
-	}
-
-	uint32 fmtLength = stream.readUint32LE();
-	if (fmtLength < 16) {
-		// A valid fmt chunk always contains at least 16 bytes
-		warning("getWavInfo: 'fmt' header is too short");
-		return false;
-	}
-
-	// Next comes the "type" field of the fmt header. Some typical
-	// values for it:
-	// 1  -> uncompressed PCM
-	// 17 -> IMA ADPCM compressed WAVE
-	// See <http://www.saettler.com/RIFFNEW/RIFFNEW.htm> for a more complete
-	// list of common WAVE compression formats...
-	uint16 type = stream.readUint16LE();	// == 1 for PCM data
-	uint16 numChannels = stream.readUint16LE();	// 1 for mono, 2 for stereo
-	uint32 samplesPerSec = stream.readUint32LE();	// in Hz
-	uint32 avgBytesPerSec = stream.readUint32LE();	// == SampleRate * NumChannels * BitsPerSample/8
-
-	uint16 blockAlign = stream.readUint16LE();	// == NumChannels * BitsPerSample/8
-	uint16 bitsPerSample = stream.readUint16LE();	// 8, 16 ...
-	// 8 bit data is unsigned, 16 bit data signed
-
-
-	if (wavType != 0)
-		*wavType = type;
-
-	if (blockAlign_ != 0)
-		*blockAlign_ = blockAlign;
-#if 0
-	debug("WAVE information:");
-	debug("  total size: %d", wavLength);
-	debug("  fmt size: %d", fmtLength);
-	debug("  type: %d", type);
-	debug("  numChannels: %d", numChannels);
-	debug("  samplesPerSec: %d", samplesPerSec);
-	debug("  avgBytesPerSec: %d", avgBytesPerSec);
-	debug("  blockAlign: %d", blockAlign);
-	debug("  bitsPerSample: %d", bitsPerSample);
-#endif
-
-	if (type != 1 && type != 2 && type != 17) {
-		warning("getWavInfo: only PCM, MS ADPCM or IMA ADPCM data is supported (type %d)", type);
-		return false;
-	}
-
-	if (blockAlign != numChannels * bitsPerSample / 8 && type != 2) {
-		debug(0, "getWavInfo: blockAlign is invalid");
-	}
-
-	if (avgBytesPerSec != samplesPerSec * blockAlign && type != 2) {
-		debug(0, "getWavInfo: avgBytesPerSec is invalid");
-	}
-
-	// Prepare the return values.
-	rate = samplesPerSec;
-
-	flags = 0;
-	if (bitsPerSample == 8)		// 8 bit data is unsigned
-		flags |= Audio::FLAG_UNSIGNED;
-	else if (bitsPerSample == 16)	// 16 bit data is signed little endian
-		flags |= (Audio::FLAG_16BITS | Audio::FLAG_LITTLE_ENDIAN);
-	else if (bitsPerSample == 4 && (type == 2 || type == 17))
-		flags |= Audio::FLAG_16BITS;
-	else {
-		warning("getWavInfo: unsupported bitsPerSample %d", bitsPerSample);
-		return false;
-	}
-
-	if (numChannels == 2)
-		flags |= Audio::FLAG_STEREO;
-	else if (numChannels != 1) {
-		warning("getWavInfo: unsupported number of channels %d", numChannels);
-		return false;
-	}
-
-	// It's almost certainly a WAV file, but we still need to find its
-	// 'data' chunk.
-
-	// Skip over the rest of the fmt chunk.
-	int offset = fmtLength - 16;
-
-	do {
-		stream.seek(offset, SEEK_CUR);
-		if (stream.pos() >= initialPos + wavLength + 8) {
-			warning("getWavInfo: Can't find 'data' chunk");
-			return false;
-		}
-		stream.read(buf, 4);
-		offset = stream.readUint32LE();
-
-#if 0
-		debug("  found a '%s' tag of size %d", buf, offset);
-#endif
-	} while (memcmp(buf, "data", 4) != 0);
-
-	// Stream now points at 'offset' bytes of sample data...
-	size = offset;
-
-	return true;
-}
-
-RewindableAudioStream *makeWAVStream(Common::SeekableReadStream *stream, DisposeAfterUse::Flag disposeAfterUse) {
-	int size, rate;
-	byte flags;
-	uint16 type;
-	int blockAlign;
-
-	if (!loadWAVFromStream(*stream, size, rate, flags, &type, &blockAlign)) {
-		if (disposeAfterUse == DisposeAfterUse::YES)
-			delete stream;
-		return 0;
-	}
-
-	if (type == 17) // MS IMA ADPCM
-		return makeADPCMStream(stream, disposeAfterUse, size, Audio::kADPCMMSIma, rate, (flags & Audio::FLAG_STEREO) ? 2 : 1, blockAlign);
-	else if (type == 2) // MS ADPCM
-		return makeADPCMStream(stream, disposeAfterUse, size, Audio::kADPCMMS, rate, (flags & Audio::FLAG_STEREO) ? 2 : 1, blockAlign);
-
-	// Raw PCM. Just read everything at once.
-	// TODO: More elegant would be to wrap the stream.
-	byte *data = (byte *)malloc(size);
-	assert(data);
-	stream->read(data, size);
-
-	if (disposeAfterUse == DisposeAfterUse::YES)
-		delete stream;
-
-	return makeRawStream(data, size, rate, flags);
-}
-
-} // End of namespace Audio
diff --git a/sound/decoders/wave.h b/sound/decoders/wave.h
deleted file mode 100644
index 2bdbe8f0b6..0000000000
--- a/sound/decoders/wave.h
+++ /dev/null
@@ -1,84 +0,0 @@
-/* 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
- *  - gob
- *  - mohawk
- *  - saga
- *  - sci
- *  - scumm
- *  - sword1
- *  - sword2
- *  - tucker
- */
-
-#ifndef SOUND_WAVE_H
-#define SOUND_WAVE_H
-
-#include "common/scummsys.h"
-#include "common/types.h"
-
-namespace Common { class SeekableReadStream; }
-
-namespace Audio {
-
-class RewindableAudioStream;
-
-/**
- * Try to load a WAVE from the given seekable stream. Returns true if
- * successful. In that case, the stream's seek position will be set to the
- * start of the audio data, and size, rate and flags contain information
- * necessary for playback. Currently this function supports uncompressed
- * raw PCM data, MS IMA ADPCM and MS ADPCM (uses makeADPCMStream internally).
- */
-extern bool loadWAVFromStream(
-	Common::SeekableReadStream &stream,
-	int &size,
-	int &rate,
-	byte &flags,
-	uint16 *wavType = 0,
-	int *blockAlign = 0);
-
-/**
- * Try to load a WAVE from the given seekable stream and create an AudioStream
- * from that data. Currently this function supports uncompressed
- * raw PCM data, MS IMA ADPCM and MS ADPCM (uses makeADPCMStream internally).
- *
- * This function uses loadWAVFromStream() internally.
- *
- * @param stream			the SeekableReadStream from which to read the WAVE data
- * @param disposeAfterUse	whether to delete the stream after use
- * @return	a new RewindableAudioStream, or NULL, if an error occurred
- */
-RewindableAudioStream *makeWAVStream(
-	Common::SeekableReadStream *stream,
-	DisposeAfterUse::Flag disposeAfterUse);
-
-} // End of namespace Audio
-
-#endif
diff --git a/sound/fmopl.cpp b/sound/fmopl.cpp
deleted file mode 100644
index ee54a79a46..0000000000
--- a/sound/fmopl.cpp
+++ /dev/null
@@ -1,197 +0,0 @@
-/* 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 "sound/fmopl.h"
-
-#include "sound/softsynth/opl/dosbox.h"
-#include "sound/softsynth/opl/mame.h"
-
-#include "common/config-manager.h"
-#include "common/translation.h"
-
-namespace OPL {
-
-// Config implementation
-
-enum OplEmulator {
-	kAuto = 0,
-	kMame = 1,
-	kDOSBox = 2
-};
-
-OPL::OPL() {
-	if (_hasInstance)
-		error("There are multiple OPL output instances running");
-	_hasInstance = true;
-}
-
-const Config::EmulatorDescription Config::_drivers[] = {
-	{ "auto", "<default>", kAuto, kFlagOpl2 | kFlagDualOpl2 | kFlagOpl3 },
-	{ "mame", _s("MAME OPL emulator"), kMame, kFlagOpl2 },
-#ifndef DISABLE_DOSBOX_OPL
-	{ "db", _s("DOSBox OPL emulator"), kDOSBox, kFlagOpl2 | kFlagDualOpl2 | kFlagOpl3 },
-#endif
-	{ 0, 0, 0, 0 }
-};
-
-Config::DriverId Config::parse(const Common::String &name) {
-	for (int i = 0; _drivers[i].name; ++i) {
-		if (name.equalsIgnoreCase(_drivers[i].name))
-			return _drivers[i].id;
-	}
-
-	return -1;
-}
-
-Config::DriverId Config::detect(OplType type) {
-	uint32 flags = 0;
-	switch (type) {
-	case kOpl2:
-		flags = kFlagOpl2;
-		break;
-
-	case kDualOpl2:
-		flags = kFlagDualOpl2;
-		break;
-
-	case kOpl3:
-		flags = kFlagOpl3;
-		break;
-	}
-
-	DriverId drv = parse(ConfMan.get("opl_driver"));
-
-	// When a valid driver is selected, check whether it supports
-	// the requested OPL chip.
-	if (drv != -1 && drv != kAuto) {
-		// If the chip is supported, just use the driver.
-		if ((flags & _drivers[drv].flags)) {
-			return drv;
-		} else {
-			// Else we will output a warning and just
-			// return that no valid driver is found.
-			warning("Your selected OPL driver \"%s\" does not support type %d emulation, which is requested by your game", _drivers[drv].description, type);
-			return -1;
-		}
-	}
-
-	// Detect the first matching emulator
-	drv = -1;
-
-	for (int i = 1; _drivers[i].name; ++i) {
-		if (_drivers[i].flags & flags) {
-			drv = _drivers[i].id;
-			break;
-		}
-	}
-
-	return drv;
-}
-
-OPL *Config::create(OplType type) {
-	return create(kAuto, type);
-}
-
-OPL *Config::create(DriverId driver, OplType type) {
-	// On invalid driver selection, we try to do some fallback detection
-	if (driver == -1) {
-		warning("Invalid OPL driver selected, trying to detect a fallback emulator");
-		driver = kAuto;
-	}
-
-	// If autodetection is selected, we search for a matching
-	// driver.
-	if (driver == kAuto) {
-		driver = detect(type);
-
-		// No emulator for the specified OPL chip could
-		// be found, thus stop here.
-		if (driver == -1) {
-			warning("No OPL emulator available for type %d", type);
-			return 0;
-		}
-	}
-
-	switch (driver) {
-	case kMame:
-		if (type == kOpl2)
-			return new MAME::OPL();
-		else
-			warning("MAME OPL emulator only supports OPL2 emulation");
-			return 0;
-
-#ifndef DISABLE_DOSBOX_OPL
-	case kDOSBox:
-		return new DOSBox::OPL(type);
-#endif
-
-	default:
-		warning("Unsupported OPL emulator %d", driver);
-		// TODO: Maybe we should add some dummy emulator too, which just outputs
-		// silence as sound?
-		return 0;
-	}
-}
-
-bool OPL::_hasInstance = false;
-
-} // End of namespace OPL
-
-void OPLDestroy(FM_OPL *OPL) {
-	delete OPL;
-}
-
-void OPLResetChip(FM_OPL *OPL) {
-	OPL->reset();
-}
-
-void OPLWrite(FM_OPL *OPL, int a, int v) {
-	OPL->write(a, v);
-}
-
-unsigned char OPLRead(FM_OPL *OPL, int a) {
-	return OPL->read(a);
-}
-
-void OPLWriteReg(FM_OPL *OPL, int r, int v) {
-	OPL->writeReg(r, v);
-}
-
-void YM3812UpdateOne(FM_OPL *OPL, int16 *buffer, int length) {
-	OPL->readBuffer(buffer, length);
-}
-
-FM_OPL *makeAdLibOPL(int rate) {
-	FM_OPL *opl = OPL::Config::create();
-
-	if (opl) {
-		if (!opl->init(rate)) {
-			delete opl;
-			opl = 0;
-		}
-	}
-
-	return opl;
-}
-
diff --git a/sound/fmopl.h b/sound/fmopl.h
deleted file mode 100644
index 33235f3545..0000000000
--- a/sound/fmopl.h
+++ /dev/null
@@ -1,179 +0,0 @@
-/* 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_FMOPL_H
-#define SOUND_FMOPL_H
-
-#include "common/scummsys.h"
-#include "common/str.h"
-
-namespace OPL {
-
-class OPL;
-
-class Config {
-public:
-	enum OplFlags {
-		kFlagOpl2		= (1 << 0),
-		kFlagDualOpl2	= (1 << 1),
-		kFlagOpl3		= (1 << 2)
-	};
-
-	/**
-	 * OPL type to emulate.
-	 */
-	enum OplType {
-		kOpl2,
-		kDualOpl2,
-		kOpl3
-	};
-
-	typedef int8 DriverId;
-	struct EmulatorDescription {
-		const char *name;
-		const char *description;
-
-		DriverId id;	// A unique ID for each driver
-		uint32 flags;	// Capabilities of this driver
-	};
-
-	/**
-	 * Get a list of all available OPL emulators.
-	 * @return list of all available OPL emulators, terminated by a zero entry
-	 */
-	static const EmulatorDescription *getAvailable() { return _drivers; }
-
-	/**
-	 * Returns the driver id of a given name.
-	 */
-	static DriverId parse(const Common::String &name);
-
-	/**
-	 * Detects a driver for the specific type.
-	 *
-	 * @return Returns a valid driver id on success, -1 otherwise.
-	 */
-	static DriverId detect(OplType type);
-
-	/**
-	 * Creates the specific driver with a specific type setup.
-	 */
-	static OPL *create(DriverId driver, OplType type);
-
-	/**
-	 * Wrapper to easily init an OPL chip, without specifing an emulator.
-	 * By default it will try to initialize an OPL2 emulator, thus an AdLib card.
-	 */
-	static OPL *create(OplType type = kOpl2);
-
-private:
-	static const EmulatorDescription _drivers[];
-};
-
-class OPL {
-private:
-	static bool _hasInstance;
-public:
-	OPL();
-	virtual ~OPL() { _hasInstance = false; }
-
-	/**
-	 * Initializes the OPL emulator.
-	 *
-	 * @param rate	output sample rate
-	 * @return		true on success, false on failure
-	 */
-	virtual bool init(int rate) = 0;
-
-	/**
-	 * Reinitializes the OPL emulator
-	 */
-	virtual void reset() = 0;
-
-	/**
-	 * Writes a byte to the given I/O port.
-	 *
-	 * @param a		port address
-	 * @param v		value, which will be written
-	 */
-	virtual void write(int a, int v) = 0;
-
-	/**
-	 * Reads a byte from the given I/O port.
-	 *
-	 * @param a		port address
-	 * @return		value read
-	 */
-	virtual byte read(int a) = 0;
-
-	/**
-	 * Function to directly write to a specific OPL register.
-	 * This writes to *both* chips for a Dual OPL2.
-	 *
-	 * @param r		hardware register number to write to
-	 * @param v		value, which will be written
-	 */
-	virtual void writeReg(int r, int v) = 0;
-
-	/**
-	 * Read up to 'length' samples.
-	 *
-	 * Data will be in native endianess, 16 bit per sample, signed.
-	 * For stereo OPL, buffer will be filled with interleaved
-	 * left and right channel samples, starting with a left sample.
-	 * Furthermore, the samples in the left and right are summed up.
-	 * So if you request 4 samples from a stereo OPL, you will get
-	 * a total of two left channel and two right channel samples.
-	 */
-	virtual void readBuffer(int16 *buffer, int length) = 0;
-
-	/**
-	 * Returns whether the setup OPL mode is stereo or not
-	 */
-	virtual bool isStereo() const = 0;
-};
-
-} // End of namespace OPL
-
-// Legacy API
-// !You should not write any new code using the legacy API!
-typedef OPL::OPL FM_OPL;
-
-void OPLDestroy(FM_OPL *OPL);
-
-void OPLResetChip(FM_OPL *OPL);
-void OPLWrite(FM_OPL *OPL, int a, int v);
-unsigned char OPLRead(FM_OPL *OPL, int a);
-void OPLWriteReg(FM_OPL *OPL, int r, int v);
-void YM3812UpdateOne(FM_OPL *OPL, int16 *buffer, int length);
-
-/**
- * Legacy factory to create an AdLib (OPL2) chip.
- *
- * !You should not write any new code using the legacy API!
- */
-FM_OPL *makeAdLibOPL(int rate);
-
-#endif
-
diff --git a/sound/mididrv.cpp b/sound/mididrv.cpp
deleted file mode 100644
index 20d5a4e233..0000000000
--- a/sound/mididrv.cpp
+++ /dev/null
@@ -1,326 +0,0 @@
-/* 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 "engines/engine.h"
-#include "common/config-manager.h"
-#include "common/str.h"
-#include "common/system.h"
-#include "common/util.h"
-#include "sound/mididrv.h"
-#include "sound/musicplugin.h"
-#include "common/translation.h"
-
-const byte MidiDriver::_mt32ToGm[128] = {
-//	  0    1    2    3    4    5    6    7    8    9    A    B    C    D    E    F
-	  0,   1,   0,   2,   4,   4,   5,   3,  16,  17,  18,  16,  16,  19,  20,  21, // 0x
-	  6,   6,   6,   7,   7,   7,   8, 112,  62,  62,  63,  63,  38,  38,  39,  39, // 1x
-	 88,  95,  52,  98,  97,  99,  14,  54, 102,  96,  53, 102,  81, 100,  14,  80, // 2x
-	 48,  48,  49,  45,  41,  40,  42,  42,  43,  46,  45,  24,  25,  28,  27, 104, // 3x
-	 32,  32,  34,  33,  36,  37,  35,  35,  79,  73,  72,  72,  74,  75,  64,  65, // 4x
-	 66,  67,  71,  71,  68,  69,  70,  22,  56,  59,  57,  57,  60,  60,  58,  61, // 5x
-	 61,  11,  11,  98,  14,   9,  14,  13,  12, 107, 107,  77,  78,  78,  76,  76, // 6x
-	 47, 117, 127, 118, 118, 116, 115, 119, 115, 112,  55, 124, 123,   0,  14, 117  // 7x
-};
-
-const byte MidiDriver::_gmToMt32[128] = {
-//	  0    1    2    3    4    5    6    7    8    9    A    B    C    D    E    F
-	  5,   1,   2,   7,   3,   5,  16,  21,  22, 101, 101,  97, 104, 103, 102,  20, // 0x
-	  8,   9,  11,  12,  14,  15,  87,  15,  59,  60,  61,  62,  67,  44,  79,  23, // 1x
-	 64,  67,  66,  70,  68,  69,  28,  31,  52,  54,  55,  56,  49,  51,  57, 112, // 2x
-	 48,  50,  45,  26,  34,  35,  45, 122,  89,  90,  94,  81,  92,  95,  24,  25, // 3x
-	 80,  78,  79,  78,  84,  85,  86,  82,  74,  72,  76,  77, 110, 107, 108,  76, // 4x
-	 47,  44, 111,  45,  44,  34,  44,  30,  32,  33,  88,  34,  35,  35,  38,  33, // 5x
-	 41,  36, 100,  37,  40,  34,  43,  40,  63,  21,  99, 105, 103,  86,  55,  84, // 6x
-	101, 103, 100, 120, 117, 113,  99, 128, 128, 128, 128, 124, 123, 128, 128, 128, // 7x
-};
-
-static const uint32 GUIOMapping[] = {
-	MT_PCSPK,		Common::GUIO_MIDIPCSPK,
-	MT_CMS,			Common::GUIO_MIDICMS,
-	MT_PCJR,		Common::GUIO_MIDIPCJR,
-	MT_ADLIB,		Common::GUIO_MIDIADLIB,
-	MT_C64,		    Common::GUIO_MIDIC64,
-	MT_AMIGA,	    Common::GUIO_MIDIAMIGA,
-	MT_APPLEIIGS,	Common::GUIO_MIDIAPPLEIIGS,
-	MT_TOWNS,		Common::GUIO_MIDITOWNS,
-	MT_PC98,		Common::GUIO_MIDIPC98,
-	MT_GM,			Common::GUIO_MIDIGM,
-	MT_MT32,		Common::GUIO_MIDIMT32,
-	0,				0
-};
-
-uint32 MidiDriver::musicType2GUIO(uint32 musicType) {
-	uint32 res = 0;
-
-	for (int i = 0; GUIOMapping[i] || GUIOMapping[i + 1]; i += 2) {
-		if (musicType == GUIOMapping[i] || musicType == (uint32)-1)
-			res |= GUIOMapping[i + 1];
-	}
-
-	return res;
-}
-
-bool MidiDriver::_forceTypeMT32 = false;
-
-MusicType MidiDriver::getMusicType(MidiDriver::DeviceHandle handle) {
-	if (_forceTypeMT32)
-		return MT_MT32;
-
-	if (handle) {
-		const MusicPlugin::List p = MusicMan.getPlugins();
-		for (MusicPlugin::List::const_iterator m = p.begin(); m != p.end(); m++) {
-			MusicDevices i = (**m)->getDevices();
-			for (MusicDevices::iterator d = i.begin(); d != i.end(); d++) {
-				if (handle == d->getHandle())
-					return d->getMusicType();
-			}
-		}
-	}
-
-	return MT_INVALID;
-}
-
-Common::String MidiDriver::getDeviceString(DeviceHandle handle, DeviceStringType type) {
-	if (handle) {
-		const MusicPlugin::List p = MusicMan.getPlugins();
-		for (MusicPlugin::List::const_iterator m = p.begin(); m != p.end(); m++) {
-			MusicDevices i = (**m)->getDevices();
-			for (MusicDevices::iterator d = i.begin(); d != i.end(); d++) {
-				if (handle == d->getHandle()) {
-					if (type == kDriverName)
-						return d->getMusicDriverName();
-					else if (type == kDriverId)
-						return d->getMusicDriverId();
-					else if (type == kDeviceId)
-						return d->getCompleteId();
-					else
-						return Common::String("auto");
-				}
-			}
-		}
-	}
-
-	return Common::String("auto");
-}
-
-MidiDriver::DeviceHandle MidiDriver::detectDevice(int flags) {
-	// Query the selected music device (defaults to MT_AUTO device).
-	DeviceHandle hdl = getDeviceHandle(ConfMan.get("music_driver"));
-
-	_forceTypeMT32 = false;
-
-	// Check whether the selected music driver is compatible with the
-	// given flags.
-	switch (getMusicType(hdl)) {
-	case MT_PCSPK:
-		if (flags & MDT_PCSPK)
-			return hdl;
-		break;
-
-	case MT_PCJR:
-		if (flags & MDT_PCJR)
-			return hdl;
-		break;
-
-	case MT_CMS:
-		if (flags & MDT_CMS)
-			return hdl;
-		break;
-
-	case MT_ADLIB:
-		if (flags & MDT_ADLIB)
-			return hdl;
-		break;
-
-	case MT_C64:
-		if (flags & MDT_C64)
-			return hdl;
-		break;
-
-	case MT_AMIGA:
-		if (flags & MDT_AMIGA)
-			return hdl;
-		break;
-
-	case MT_APPLEIIGS:
-		if (flags & MDT_APPLEIIGS)
-			return hdl;
-		break;
-
-	case MT_TOWNS:
-		if (flags & MDT_TOWNS)
-			return hdl;
-		break;
-
-	case MT_PC98:
-		if (flags & MDT_PC98)
-			return hdl;
-		break;
-
-	case MT_GM:
-	case MT_GS:
-	case MT_MT32:
-		if (flags & MDT_MIDI)
-			return hdl;
-		break;
-
-	case MT_NULL:
-		return hdl;
-
-	default:
-		break;
-	}
-
-	// If the selected driver did not match the flags setting,
-	// we try to determine a suitable and "optimal" music driver.
-	const MusicPlugin::List p = MusicMan.getPlugins();
-	// If only MDT_MIDI but not MDT_PREFER_MT32 or MDT_PREFER_GM is set we prefer the other devices (which will always be
-	// detected since they are hard coded and cannot be disabled.
-	for (int l = (flags & (MDT_PREFER_GM | MDT_PREFER_MT32)) ? 1 : 0; l < 2; ++l) {
-		if ((flags & MDT_MIDI) && (l == 1)) {
-			// If a preferred MT32 or GM device has been selected that device gets returned
-			if (flags & MDT_PREFER_MT32)
-				hdl = getDeviceHandle(ConfMan.get("mt32_device"));
-			else if (flags & MDT_PREFER_GM)
-				hdl = getDeviceHandle(ConfMan.get("gm_device"));
-			else
-				hdl = getDeviceHandle("auto");
-
-			const MusicType type = getMusicType(hdl);
-
-			// If have a "Don't use GM/MT-32" setting we skip this part and jump
-			// to AdLib, PC Speaker etc. detection right away.
-			if (type != MT_NULL) {
-				if (type != MT_AUTO && type != MT_INVALID) {
-					if (flags & MDT_PREFER_MT32)
-						// If we have a preferred MT32 device we disable the gm/mt32 mapping (more about this in mididrv.h)
-						_forceTypeMT32 = true;
-
-					return hdl;
-				}
-
-				// If we have no specific device selected (neither in the scummvm nor in the game domain)
-				// and no preferred MT32 or GM device selected we arrive here.
-				// If MT32 is preferred we try for the first available device with music type 'MT_MT32' (usually the mt32 emulator)
-				if (flags & MDT_PREFER_MT32) {
-					for (MusicPlugin::List::const_iterator m = p.begin(); m != p.end(); ++m) {
-						MusicDevices i = (**m)->getDevices();
-						for (MusicDevices::iterator d = i.begin(); d != i.end(); ++d) {
-							if (d->getMusicType() == MT_MT32)
-								return d->getHandle();
-						}
-					}
-				}
-
-				// Now we default to the first available device with music type 'MT_GM'
-				for (MusicPlugin::List::const_iterator m = p.begin(); m != p.end(); ++m) {
-					MusicDevices i = (**m)->getDevices();
-					for (MusicDevices::iterator d = i.begin(); d != i.end(); ++d) {
-						if (d->getMusicType() == MT_GM || d->getMusicType() == MT_GS)
-							return d->getHandle();
-					}
-				}
-			}
-		}
-
-		MusicType tp = MT_AUTO;
-		if (flags & MDT_TOWNS)
-			tp = MT_TOWNS;
-		else if (flags & MDT_PC98)
-			tp = MT_PC98;
-		else if (flags & MDT_ADLIB)
-			tp = MT_ADLIB;
-		else if (flags & MDT_PCJR)
-			tp = MT_PCJR;
-		else if (flags & MDT_PCSPK)
-			tp = MT_PCSPK;
-		else if (flags & MDT_C64)
-			tp = MT_C64;
-		else if (flags & MDT_AMIGA)
-			tp = MT_AMIGA;
-		else if (flags & MDT_APPLEIIGS)
-			tp = MT_APPLEIIGS;
-		else if (l == 0)
-			// if we haven't tried to find a MIDI device yet we do this now.
-			continue;
-		else
-			tp = MT_AUTO;
-
-		for (MusicPlugin::List::const_iterator m = p.begin(); m != p.end(); ++m) {
-			MusicDevices i = (**m)->getDevices();
-			for (MusicDevices::iterator d = i.begin(); d != i.end(); ++d) {
-				if (d->getMusicType() == tp)
-					return d->getHandle();
-			}
-		}
-	}
-
-	return 0;
-}
-
-MidiDriver *MidiDriver::createMidi(MidiDriver::DeviceHandle handle) {
-	MidiDriver *driver = 0;
-	const MusicPlugin::List p = MusicMan.getPlugins();
-	for (MusicPlugin::List::const_iterator m = p.begin(); m != p.end(); m++) {
-		if (getDeviceString(handle, MidiDriver::kDriverId).equals((**m)->getId()))
-			(**m)->createInstance(&driver, handle);
-	}
-
-	return driver;
-}
-
-MidiDriver::DeviceHandle MidiDriver::getDeviceHandle(const Common::String &identifier) {
-	const MusicPlugin::List p = MusicMan.getPlugins();
-
-	if (p.begin() == p.end())
-		error("Music plugins must be loaded prior to calling this method");
-
-	for (MusicPlugin::List::const_iterator m = p.begin(); m != p.end(); m++) {
-		MusicDevices i = (**m)->getDevices();
-		for (MusicDevices::iterator d = i.begin(); d != i.end(); d++) {
-			// The music driver id isn't unique, but it will match
-			// driver's first device. This is useful when selecting
-			// the driver from the command line.
-			if (identifier.equals(d->getMusicDriverId()) || identifier.equals(d->getCompleteId()) || identifier.equals(d->getCompleteName())) {
-				return d->getHandle();
-			}
-		}
-	}
-
-	return 0;
-}
-
-void MidiDriver::sendMT32Reset() {
-	static const byte resetSysEx[] = { 0x41, 0x10, 0x16, 0x12, 0x7F, 0x00, 0x00, 0x01, 0x00 };
-	sysEx(resetSysEx, sizeof(resetSysEx));
-	g_system->delayMillis(100);
-}
-
-void MidiDriver::sendGMReset() {
-	static const byte resetSysEx[] = { 0x7E, 0x7F, 0x09, 0x01 };
-	sysEx(resetSysEx, sizeof(resetSysEx));
-	g_system->delayMillis(100);
-}
-
diff --git a/sound/mididrv.h b/sound/mididrv.h
deleted file mode 100644
index 9e649cba3d..0000000000
--- a/sound/mididrv.h
+++ /dev/null
@@ -1,288 +0,0 @@
-/* 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_MIDIDRV_H
-#define SOUND_MIDIDRV_H
-
-#include "common/scummsys.h"
-#include "common/timer.h"
-
-class MidiChannel;
-class MusicDevice;
-
-namespace Audio {
-	class Mixer;
-}
-namespace Common { class String; }
-
-/**
- * Music Driver Types, used to uniquely identify each music driver.
- *
- * The pseudo drivers are listed first, then all native drivers,
- * then all other MIDI drivers, and finally the non-MIDI drivers.
- *
- * @todo Rename MidiDriverType to MusicDriverType
- */
-
-/**
- * Music types that music drivers can implement and engines can rely on.
- */
-enum MusicType {
-	MT_INVALID = -1,	// Invalid output
-	MT_AUTO = 0,		// Auto
-	MT_NULL,			// Null
-	MT_PCSPK,			// PC Speaker
-	MT_PCJR,			// PCjr
-	MT_CMS,				// CMS
-	MT_ADLIB,			// AdLib
-	MT_C64,				// C64
-	MT_AMIGA,			// Amiga
-	MT_APPLEIIGS,		// Apple IIGS
-	MT_TOWNS,			// FM-TOWNS
-	MT_PC98,			// PC98
-	MT_GM,				// General MIDI
-	MT_MT32,			// MT-32
-	MT_GS				// Roland GS
-};
-
-/**
- * A set of flags to be passed to detectDevice() which can be used to
- * specify what kind of music driver is preferred / accepted.
- *
- * The flags (except for MDT_PREFER_MT32 and MDT_PREFER_GM) indicate whether a given driver
- * type is acceptable. E.g. the TOWNS music driver could be returned by
- * detectDevice if and only if MDT_TOWNS is specified.
- *
- * @todo Rename MidiDriverFlags to MusicDriverFlags
- */
-enum MidiDriverFlags {
-	MDT_NONE        = 0,
-	MDT_PCSPK       = 1 << 0,		// PC Speaker: Maps to MD_PCSPK and MD_PCJR
-	MDT_CMS         = 1 << 1,		// Creative Music System / Gameblaster: Maps to MD_CMS
-	MDT_PCJR        = 1 << 2,		// Tandy/PC Junior driver
-	MDT_ADLIB       = 1 << 3,		// AdLib: Maps to MT_ADLIB
-	MDT_C64         = 1 << 4,
-	MDT_AMIGA       = 1 << 5,
-	MDT_APPLEIIGS   = 1 << 6,
-	MDT_TOWNS       = 1 << 7,		// FM-TOWNS: Maps to MT_TOWNS
-	MDT_PC98        = 1 << 8,		// FM-TOWNS: Maps to MT_PC98
-	MDT_MIDI        = 1 << 9,		// Real MIDI
-	MDT_PREFER_MT32 = 1 << 10,		// MT-32 output is preferred
-	MDT_PREFER_GM   = 1 << 11		// GM output is preferred
-};
-
-/**
- * Abstract description of a MIDI driver. Used by the config file and command
- * line parsing code, and also to be able to give the user a list of available
- * drivers.
- *
- * @todo Rename MidiDriverType to MusicDriverType
- */
-
-/**
- * Abstract MIDI Driver Class
- *
- * @todo Rename MidiDriver to MusicDriver
- */
-class MidiDriver {
-public:
-	/**
-	 * The device handle.
-	 *
-	 * The value 0 is reserved for an invalid device for now.
-	 * TODO: Maybe we should use -1 (i.e. 0xFFFFFFFF) as
-	 * invalid device?
-	 */
-	typedef uint32 DeviceHandle;
-
-	enum DeviceStringType {
-		kDriverName,
-		kDriverId,
-		kDeviceId
-	};
-
-	static uint32 musicType2GUIO(uint32 musicType);
-
-	/** Create music driver matching the given device handle, or NULL if there is no match. */
-	static MidiDriver *createMidi(DeviceHandle handle);
-
-	/** Returns device handle based on the present devices and the flags parameter. */
-	static DeviceHandle detectDevice(int flags);
-
-	/** Find the music driver matching the given driver name/description. */
-	static DeviceHandle getDeviceHandle(const Common::String &identifier);
-
-	/** Get the music type matching the given device handle, or MT_AUTO if there is no match. */
-	static MusicType getMusicType(DeviceHandle handle);
-
-	/** Get the device description string matching the given device handle and the given type. */
-	static Common::String getDeviceString(DeviceHandle handle, DeviceStringType type);
-
-private:
-	// If detectDevice() detects MT32 and we have a preferred MT32 device
-	// we use this to force getMusicType() to return MT_MT32 so that we don't
-	// have to rely on the 'True Roland MT-32' config manager setting (since nobody
-	// would possibly think about activating 'True Roland MT-32' when he has set
-	// 'Music Driver' to '<default>')
-	static bool _forceTypeMT32;
-
-public:
-	virtual ~MidiDriver() { }
-
-	static const byte _mt32ToGm[128];
-	static const byte _gmToMt32[128];
-
-	/**
-	 * Error codes returned by open.
-	 * Can be converted to a string with getErrorName().
-	 */
-	enum {
-		MERR_CANNOT_CONNECT = 1,
-//		MERR_STREAMING_NOT_AVAILABLE = 2,
-		MERR_DEVICE_NOT_AVAILABLE = 3,
-		MERR_ALREADY_OPEN = 4
-	};
-
-	enum {
-//		PROP_TIMEDIV = 1,
-		PROP_OLD_ADLIB = 2,
-		PROP_CHANNEL_MASK = 3
-	};
-
-	/**
-	 * Open the midi driver.
-	 * @return 0 if successful, otherwise an error code.
-	 */
-	virtual int open() = 0;
-
-	/** Close the midi driver. */
-	virtual void close() = 0;
-
-	/**
-	 * Output a packed midi command to the midi stream.
-	 * The 'lowest' byte (i.e. b & 0xFF) is the status
-	 * code, then come (if used) the first and second
-	 * opcode.
-	 */
-	virtual void send(uint32 b) = 0;
-
-	/**
-	 * Output a midi command to the midi stream. Convenience wrapper
-	 * around the usual 'packed' send method.
-	 *
-	 * Do NOT use this for sysEx transmission; instead, use the sysEx()
-	 * method below.
-	 */
-	void send(byte status, byte firstOp, byte secondOp) {
-		send(status | ((uint32)firstOp << 8) | ((uint32)secondOp << 16));
-	}
-
-	/** Get or set a property. */
-	virtual uint32 property(int prop, uint32 param) { return 0; }
-
-	/** Retrieve a string representation of an error code. */
-	static const char *getErrorName(int error_code);
-
-	// HIGH-LEVEL SEMANTIC METHODS
-	virtual void setPitchBendRange(byte channel, uint range) {
-		send(0xB0 | channel, 101, 0);
-		send(0xB0 | channel, 100, 0);
-		send(0xB0 | channel,   6, range);
-		send(0xB0 | channel,  38, 0);
-		send(0xB0 | channel, 101, 127);
-		send(0xB0 | channel, 100, 127);
-	}
-
-	/**
-	 * Send a Roland MT-32 reset sysEx to the midi device.
-	 */
-	void sendMT32Reset();
-
-	/**
-	 * Send a General MIDI reset sysEx to the midi device.
-	 */
-	void sendGMReset();
-
-	/**
-	 * Transmit a sysEx to the midi device.
-	 *
-	 * The given msg MUST NOT contain the usual SysEx frame, i.e.
-	 * do NOT include the leading 0xF0 and the trailing 0xF7.
-	 *
-	 * Furthermore, the maximal supported length of a SysEx
-	 * is 264 bytes. Passing longer buffers can lead to
-	 * undefined behavior (most likely, a crash).
-	 */
-	virtual void sysEx(const byte *msg, uint16 length) { }
-
-	virtual void sysEx_customInstrument(byte channel, uint32 type, const byte *instr) { }
-
-	virtual void metaEvent(byte type, byte *data, uint16 length) { }
-
-	// Timing functions - MidiDriver now operates timers
-	virtual void setTimerCallback(void *timer_param, Common::TimerManager::TimerProc timer_proc) = 0;
-
-	/** The time in microseconds between invocations of the timer callback. */
-	virtual uint32 getBaseTempo() = 0;
-
-	// Channel allocation functions
-	virtual MidiChannel *allocateChannel() = 0;
-	virtual MidiChannel *getPercussionChannel() = 0;
-};
-
-class MidiChannel {
-public:
-	virtual ~MidiChannel() {}
-
-	virtual MidiDriver *device() = 0;
-	virtual byte getNumber() = 0;
-	virtual void release() = 0;
-
-	virtual void send(uint32 b) = 0; // 4-bit channel portion is ignored
-
-	// Regular messages
-	virtual void noteOff(byte note) = 0;
-	virtual void noteOn(byte note, byte velocity) = 0;
-	virtual void programChange(byte program) = 0;
-	virtual void pitchBend(int16 bend) = 0; // -0x2000 to +0x1FFF
-
-	// Control Change messages
-	virtual void controlChange(byte control, byte value) = 0;
-	virtual void modulationWheel(byte value) { controlChange(1, value); }
-	virtual void volume(byte value) { controlChange(7, value); }
-	virtual void panPosition(byte value) { controlChange(10, value); }
-	virtual void pitchBendFactor(byte value) = 0;
-	virtual void detune(byte value) { controlChange(17, value); }
-	virtual void priority(byte value) { }
-	virtual void sustain(bool value) { controlChange(64, value ? 1 : 0); }
-	virtual void effectLevel(byte value) { controlChange(91, value); }
-	virtual void chorusLevel(byte value) { controlChange(93, value); }
-	virtual void allNotesOff() { controlChange(123, 0); }
-
-	// SysEx messages
-	virtual void sysEx_customInstrument(uint32 type, const byte *instr) = 0;
-};
-
-#endif
diff --git a/sound/midiparser.cpp b/sound/midiparser.cpp
deleted file mode 100644
index bd51a96d46..0000000000
--- a/sound/midiparser.cpp
+++ /dev/null
@@ -1,467 +0,0 @@
-/* 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 "sound/midiparser.h"
-#include "sound/mididrv.h"
-#include "common/util.h"
-
-//////////////////////////////////////////////////
-//
-// MidiParser implementation
-//
-//////////////////////////////////////////////////
-
-MidiParser::MidiParser() :
-_hanging_notes_count(0),
-_driver(0),
-_timer_rate(0x4A0000),
-_ppqn(96),
-_tempo(500000),
-_psec_per_tick(5208), // 500000 / 96
-_autoLoop(false),
-_smartJump(false),
-_centerPitchWheelOnUnload(false),
-_sendSustainOffOnNotesOff(false),
-_num_tracks(0),
-_active_track(255),
-_abort_parse(0) {
-	memset(_active_notes, 0, sizeof(_active_notes));
-	_next_event.start = NULL;
-	_next_event.delta = 0;
-	_next_event.event = 0;
-	_next_event.length = 0;
-}
-
-void MidiParser::property(int prop, int value) {
-	switch (prop) {
-	case mpAutoLoop:
-		_autoLoop = (value != 0);
-		break;
-	case mpSmartJump:
-		_smartJump = (value != 0);
-		break;
-	case mpCenterPitchWheelOnUnload:
-		_centerPitchWheelOnUnload = (value != 0);
-		break;
-	case mpSendSustainOffOnNotesOff:
-		_sendSustainOffOnNotesOff = (value != 0);
-		break;
-	}
-}
-
-void MidiParser::sendToDriver(uint32 b) {
-	_driver->send(b);
-}
-
-void MidiParser::setTempo(uint32 tempo) {
-	_tempo = tempo;
-	if (_ppqn)
-		_psec_per_tick = (tempo + (_ppqn >> 2)) / _ppqn;
-}
-
-// This is the conventional (i.e. SMF) variable length quantity
-uint32 MidiParser::readVLQ(byte * &data) {
-	byte str;
-	uint32 value = 0;
-	int i;
-
-	for (i = 0; i < 4; ++i) {
-		str = data[0];
-		++data;
-		value = (value << 7) | (str & 0x7F);
-		if (!(str & 0x80))
-			break;
-	}
-	return value;
-}
-
-void MidiParser::activeNote(byte channel, byte note, bool active) {
-	if (note >= 128 || channel >= 16)
-		return;
-
-	if (active)
-		_active_notes[note] |= (1 << channel);
-	else
-		_active_notes[note] &= ~(1 << channel);
-
-	// See if there are hanging notes that we can cancel
-	NoteTimer *ptr = _hanging_notes;
-	int i;
-	for (i = ARRAYSIZE(_hanging_notes); i; --i, ++ptr) {
-		if (ptr->channel == channel && ptr->note == note && ptr->time_left) {
-			ptr->time_left = 0;
-			--_hanging_notes_count;
-			break;
-		}
-	}
-}
-
-void MidiParser::hangingNote(byte channel, byte note, uint32 time_left, bool recycle) {
-	NoteTimer *best = 0;
-	NoteTimer *ptr = _hanging_notes;
-	int i;
-
-	if (_hanging_notes_count >= ARRAYSIZE(_hanging_notes)) {
-		warning("MidiParser::hangingNote(): Exceeded polyphony");
-		return;
-	}
-
-	for (i = ARRAYSIZE(_hanging_notes); i; --i, ++ptr) {
-		if (ptr->channel == channel && ptr->note == note) {
-			if (ptr->time_left && ptr->time_left < time_left && recycle)
-				return;
-			best = ptr;
-			if (ptr->time_left) {
-				if (recycle)
-					sendToDriver(0x80 | channel, note, 0);
-				--_hanging_notes_count;
-			}
-			break;
-		} else if (!best && ptr->time_left == 0) {
-			best = ptr;
-		}
-	}
-
-	// Occassionally we might get a zero or negative note
-	// length, if the note should be turned on and off in
-	// the same iteration. For now just set it to 1 and
-	// we'll turn it off in the next cycle.
-	if (!time_left || time_left & 0x80000000)
-		time_left = 1;
-
-	if (best) {
-		best->channel = channel;
-		best->note = note;
-		best->time_left = time_left;
-		++_hanging_notes_count;
-	} else {
-		// We checked this up top. We should never get here!
-		warning("MidiParser::hangingNote(): Internal error");
-	}
-}
-
-void MidiParser::onTimer() {
-	uint32 end_time;
-	uint32 event_time;
-
-	if (!_position._play_pos || !_driver)
-		return;
-
-	_abort_parse = false;
-	end_time = _position._play_time + _timer_rate;
-
-	// Scan our hanging notes for any
-	// that should be turned off.
-	if (_hanging_notes_count) {
-		NoteTimer *ptr = &_hanging_notes[0];
-		int i;
-		for (i = ARRAYSIZE(_hanging_notes); i; --i, ++ptr) {
-			if (ptr->time_left) {
-				if (ptr->time_left <= _timer_rate) {
-					sendToDriver(0x80 | ptr->channel, ptr->note, 0);
-					ptr->time_left = 0;
-					--_hanging_notes_count;
-				} else {
-					ptr->time_left -= _timer_rate;
-				}
-			}
-		}
-	}
-
-	while (!_abort_parse) {
-		EventInfo &info = _next_event;
-
-		event_time = _position._last_event_time + info.delta * _psec_per_tick;
-		if (event_time > end_time)
-			break;
-
-		// Process the next info.
-		_position._last_event_tick += info.delta;
-		if (info.event < 0x80) {
-			warning("Bad command or running status %02X", info.event);
-			_position._play_pos = 0;
-			return;
-		}
-
-		if (info.event == 0xF0) {
-			// SysEx event
-			// Check for trailing 0xF7 -- if present, remove it.
-			if (info.ext.data[info.length-1] == 0xF7)
-				_driver->sysEx(info.ext.data, (uint16)info.length-1);
-			else
-				_driver->sysEx(info.ext.data, (uint16)info.length);
-		} else if (info.event == 0xFF) {
-			// META event
-			if (info.ext.type == 0x2F) {
-				// End of Track must be processed by us,
-				// as well as sending it to the output device.
-				if (_autoLoop) {
-					jumpToTick(0);
-					parseNextEvent(_next_event);
-				} else {
-					stopPlaying();
-					_driver->metaEvent(info.ext.type, info.ext.data, (uint16)info.length);
-				}
-				return;
-			} else if (info.ext.type == 0x51) {
-				if (info.length >= 3) {
-					setTempo(info.ext.data[0] << 16 | info.ext.data[1] << 8 | info.ext.data[2]);
-				}
-			}
-			_driver->metaEvent(info.ext.type, info.ext.data, (uint16)info.length);
-		} else {
-			if (info.command() == 0x8) {
-				activeNote(info.channel(), info.basic.param1, false);
-			} else if (info.command() == 0x9) {
-				if (info.length > 0)
-					hangingNote(info.channel(), info.basic.param1, info.length * _psec_per_tick - (end_time - event_time));
-				else
-					activeNote(info.channel(), info.basic.param1, true);
-			}
-			sendToDriver(info.event, info.basic.param1, info.basic.param2);
-		}
-
-
-		if (!_abort_parse) {
-			_position._last_event_time = event_time;
-			parseNextEvent(_next_event);
-		}
-	}
-
-	if (!_abort_parse) {
-		_position._play_time = end_time;
-		_position._play_tick = (_position._play_time - _position._last_event_time) / _psec_per_tick + _position._last_event_tick;
-	}
-}
-
-void MidiParser::allNotesOff() {
-	if (!_driver)
-		return;
-
-	int i, j;
-
-	// Turn off all active notes
-	for (i = 0; i < 128; ++i) {
-		for (j = 0; j < 16; ++j) {
-			if (_active_notes[i] & (1 << j)) {
-				sendToDriver(0x80 | j, i, 0);
-			}
-		}
-	}
-
-	// Turn off all hanging notes
-	for (i = 0; i < ARRAYSIZE(_hanging_notes); i++) {
-		if (_hanging_notes[i].time_left) {
-			sendToDriver(0x80 | _hanging_notes[i].channel, _hanging_notes[i].note, 0);
-			_hanging_notes[i].time_left = 0;
-		}
-	}
-	_hanging_notes_count = 0;
-
-	// To be sure, send an "All Note Off" event (but not all MIDI devices
-	// support this...).
-
-	for (i = 0; i < 16; ++i) {
-		sendToDriver(0xB0 | i, 0x7b, 0); // All notes off
-		if (_sendSustainOffOnNotesOff)
-			sendToDriver(0xB0 | i, 0x40, 0); // Also send a sustain off event (bug #3116608)
-	}
-
-	memset(_active_notes, 0, sizeof(_active_notes));
-}
-
-void MidiParser::resetTracking() {
-	_position.clear();
-}
-
-bool MidiParser::setTrack(int track) {
-	if (track < 0 || track >= _num_tracks)
-		return false;
-	// We allow restarting the track via setTrack when
-	// it isn't playing anymore. This allows us to reuse
-	// a MidiParser when a track has finished and will
-	// be restarted via setTrack by the client again.
-	// This isn't exactly how setTrack behaved before though,
-	// the old MidiParser code did not allow setTrack to be
-	// used to restart a track, which was already finished.
-	//
-	// TODO: Check if any engine has problem with this
-	// handling, if so we need to find a better way to handle
-	// track restarts. (KYRA relies on this working)
-	else if (track == _active_track && isPlaying())
-		return true;
-
-	if (_smartJump)
-		hangAllActiveNotes();
-	else
-		allNotesOff();
-
-	resetTracking();
-	memset(_active_notes, 0, sizeof(_active_notes));
-	_active_track = track;
-	_position._play_pos = _tracks[track];
-	parseNextEvent(_next_event);
-	return true;
-}
-
-void MidiParser::stopPlaying() {
-	allNotesOff();
-	resetTracking();
-}
-
-void MidiParser::hangAllActiveNotes() {
-	// Search for note off events until we have
-	// accounted for every active note.
-	uint16 temp_active[128];
-	memcpy(temp_active, _active_notes, sizeof (temp_active));
-
-	uint32 advance_tick = _position._last_event_tick;
-	while (true) {
-		int i;
-		for (i = 0; i < 128; ++i)
-			if (temp_active[i] != 0)
-				break;
-		if (i == 128)
-			break;
-		parseNextEvent(_next_event);
-		advance_tick += _next_event.delta;
-		if (_next_event.command() == 0x8) {
-			if (temp_active[_next_event.basic.param1] & (1 << _next_event.channel())) {
-				hangingNote(_next_event.channel(), _next_event.basic.param1, (advance_tick - _position._last_event_tick) * _psec_per_tick, false);
-				temp_active[_next_event.basic.param1] &= ~ (1 << _next_event.channel());
-			}
-		} else if (_next_event.event == 0xFF && _next_event.ext.type == 0x2F) {
-			// warning("MidiParser::hangAllActiveNotes(): Hit End of Track with active notes left");
-			for (i = 0; i < 128; ++i) {
-				for (int j = 0; j < 16; ++j) {
-					if (temp_active[i] & (1 << j)) {
-						activeNote(j, i, false);
-						sendToDriver(0x80 | j, i, 0);
-					}
-				}
-			}
-			break;
-		}
-	}
-}
-
-bool MidiParser::jumpToTick(uint32 tick, bool fireEvents, bool stopNotes, bool dontSendNoteOn) {
-	if (_active_track >= _num_tracks)
-		return false;
-
-	Tracker currentPos(_position);
-	EventInfo currentEvent(_next_event);
-
-	resetTracking();
-	_position._play_pos = _tracks[_active_track];
-	parseNextEvent(_next_event);
-	if (tick > 0) {
-		while (true) {
-			EventInfo &info = _next_event;
-			if (_position._last_event_tick + info.delta >= tick) {
-				_position._play_time += (tick - _position._last_event_tick) * _psec_per_tick;
-				_position._play_tick = tick;
-				break;
-			}
-
-			_position._last_event_tick += info.delta;
-			_position._last_event_time += info.delta * _psec_per_tick;
-			_position._play_tick = _position._last_event_tick;
-			_position._play_time = _position._last_event_time;
-
-			if (info.event == 0xFF) {
-				if (info.ext.type == 0x2F) { // End of track
-					_position = currentPos;
-					_next_event = currentEvent;
-					return false;
-				} else {
-					if (info.ext.type == 0x51 && info.length >= 3) // Tempo
-						setTempo(info.ext.data[0] << 16 | info.ext.data[1] << 8 | info.ext.data[2]);
-					if (fireEvents)
-						_driver->metaEvent(info.ext.type, info.ext.data, (uint16) info.length);
-				}
-			} else if (fireEvents) {
-				if (info.event == 0xF0) {
-					if (info.ext.data[info.length-1] == 0xF7)
-						_driver->sysEx(info.ext.data, (uint16)info.length-1);
-					else
-						_driver->sysEx(info.ext.data, (uint16)info.length);
-				} else {
-					// The note on sending code is used by the SCUMM engine. Other engine using this code
-					// (such as SCI) have issues with this, as all the notes sent can be heard when a song
-					// is fast-forwarded.	Thus, if the engine requests it, don't send note on events.
-					if (info.command() == 0x9 && dontSendNoteOn) {
-						// Don't send note on; doing so creates a "warble" with some instruments on the MT-32.
-						// Refer to patch #3117577
-					} else {
-						sendToDriver(info.event, info.basic.param1, info.basic.param2);
-					}
-				}
-			}
-
-			parseNextEvent(_next_event);
-		}
-	}
-
-	if (stopNotes) {
-		if (!_smartJump || !currentPos._play_pos) {
-			allNotesOff();
-		} else {
-			EventInfo targetEvent(_next_event);
-			Tracker targetPosition(_position);
-
-			_position = currentPos;
-			_next_event = currentEvent;
-			hangAllActiveNotes();
-
-			_next_event = targetEvent;
-			_position = targetPosition;
-		}
-	}
-
-	_abort_parse = true;
-	return true;
-}
-
-void MidiParser::unloadMusic() {
-	resetTracking();
-	allNotesOff();
-	_num_tracks = 0;
-	_active_track = 255;
-	_abort_parse = true;
-
-	if (_centerPitchWheelOnUnload) {
-		// Center the pitch wheels in preparation for the next piece of
-		// music. It's not safe to do this from within allNotesOff(),
-		// and might not even be safe here, so we only do it if the
-		// client has explicitly asked for it.
-
-		if (_driver) {
-			for (int i = 0; i < 16; ++i) {
-				sendToDriver(0xE0 | i, 0, 0x40);
-			}
-		}
-	}
-}
diff --git a/sound/midiparser.h b/sound/midiparser.h
deleted file mode 100644
index 0b18a19a5b..0000000000
--- a/sound/midiparser.h
+++ /dev/null
@@ -1,404 +0,0 @@
-/* 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.
- *
- * $URL$
- * $Id$
- *
- */
-
-/// \brief Declarations related to the MidiParser class
-
-#ifndef SOUND_MIDIPARSER_H
-#define SOUND_MIDIPARSER_H
-
-#include "common/scummsys.h"
-#include "common/endian.h"
-
-class MidiParser;
-class MidiDriver;
-
-
-
-//////////////////////////////////////////////////
-//
-// Support entities
-//
-//////////////////////////////////////////////////
-
-/**
- * Maintains time and position state within a MIDI stream.
- * A single Tracker struct is used by MidiParser to keep track
- * of its current position in the MIDI stream. The Tracker
- * struct, however, allows alternative locations to be cached.
- * See MidiParser::jumpToTick() for an example of tracking
- * multiple locations within a MIDI stream. NOTE: It is
- * important to also maintain pre-parsed EventInfo data for
- * each Tracker location.
- */
-struct Tracker {
-	byte * _play_pos;        ///< A pointer to the next event to be parsed
-	uint32 _play_time;       ///< Current time in microseconds; may be in between event times
-	uint32 _play_tick;       ///< Current MIDI tick; may be in between event ticks
-	uint32 _last_event_time; ///< The time, in microseconds, of the last event that was parsed
-	uint32 _last_event_tick; ///< The tick at which the last parsed event occurs
-	byte   _running_status;  ///< Cached MIDI command, for MIDI streams that rely on implied event codes
-
-	Tracker() { clear(); }
-
-	/// Copy constructor for each duplication of Tracker information.
-	Tracker(const Tracker &copy) :
-	_play_pos(copy._play_pos),
-	_play_time(copy._play_time),
-	_play_tick(copy._play_tick),
-	_last_event_time(copy._last_event_time),
-	_last_event_tick(copy._last_event_tick),
-	_running_status(copy._running_status)
-	{ }
-
-	/// Clears all data; used by the constructor for initialization.
-	void clear() {
-		_play_pos = 0;
-		_play_time = 0;
-		_play_tick = 0;
-		_last_event_time = 0;
-		_last_event_tick = 0;
-		_running_status = 0;
-	}
-};
-
-/**
- * Provides comprehensive information on the next event in the MIDI stream.
- * An EventInfo struct is instantiated by format-specific implementations
- * of MidiParser::parseNextEvent() each time another event is needed.
- */
-struct EventInfo {
-	byte * start; ///< Position in the MIDI stream where the event starts.
-	              ///< For delta-based MIDI streams (e.g. SMF and XMIDI), this points to the delta.
-	uint32 delta; ///< The number of ticks after the previous event that this event should occur.
-	byte   event; ///< Upper 4 bits are the command code, lower 4 bits are the MIDI channel.
-	              ///< For META, event == 0xFF. For SysEx, event == 0xF0.
-	union {
-		struct {
-			byte param1; ///< The first parameter in a simple MIDI message.
-			byte param2; ///< The second parameter in a simple MIDI message.
-		} basic;
-		struct {
-			byte   type; ///< For META events, this indicates the META type.
-			byte * data; ///< For META and SysEx events, this points to the start of the data.
-		} ext;
-	};
-	uint32 length; ///< For META and SysEx blocks, this indicates the length of the data.
-	               ///< For Note On events, a non-zero value indicates that no Note Off event
-	               ///< will occur, and the MidiParser will have to generate one itself.
-	               ///< For all other events, this value should always be zero.
-
-	byte channel() { return event & 0x0F; } ///< Separates the MIDI channel from the event.
-	byte command() { return event >> 4; }   ///< Separates the command code from the event.
-};
-
-/**
- * Provides expiration tracking for hanging notes.
- * Hanging notes are used when a MIDI format does not include explicit Note Off
- * events, or when "Smart Jump" is enabled so that active notes are intelligently
- * expired when a jump occurs. The NoteTimer struct keeps track of how much
- * longer a note should remain active before being turned off.
- */
-struct NoteTimer {
-	byte channel;     ///< The MIDI channel on which the note was played
-	byte note;        ///< The note number for the active note
-	uint32 time_left; ///< The time, in microseconds, remaining before the note should be turned off
-	NoteTimer() : channel(0), note(0), time_left(0) {}
-};
-
-
-
-
-//////////////////////////////////////////////////
-//
-// MidiParser declaration
-//
-//////////////////////////////////////////////////
-
-/**
- * A framework and common functionality for parsing event-based music streams.
- * The MidiParser provides a framework in which to load,
- * parse and traverse event-based music data. Note the
- * avoidance of the phrase "MIDI data." Despite its name,
- * MidiParser derivatives can be used to manage a wide
- * variety of event-based music formats. It is, however,
- * based on the premise that the format in question can
- * be played in the form of specification MIDI events.
- *
- * In order to use MidiParser to parse your music format,
- * follow these steps:
- *
- * <b>STEP 1: Write a MidiParser derivative.</b>
- * The MidiParser base class provides functionality
- * considered common to the task of parsing event-based
- * music. In order to parse a particular format, create
- * a derived class that implements, at minimum, the
- * following format-specific methods:
- *   - loadMusic
- *   - parseNextEvent
- *
- * In addition to the above functions, the derived class
- * may also override the default MidiParser behavior for
- * the following methods:
- *   - resetTracking
- *   - allNotesOff
- *   - unloadMusic
- *   - property
- *   - getTick
- *
- * Please see the documentation for these individual
- * functions for more information on their use.
- *
- * The naming convention for classes derived from
- * MidiParser is MidiParser_XXX, where "XXX" is some
- * short designator for the format the class will
- * support. For instance, the MidiParser derivative
- * for parsing the Standard MIDI File format is
- * MidiParser_SMF.
- *
- * <b>STEP 2: Create an object of your derived class.</b>
- * Each MidiParser object can parse at most one (1) song
- * at a time. However, a MidiParser object can be reused
- * to play another song once it is no longer needed to
- * play whatever it was playing. In other words, MidiParser
- * objects do not have to be destroyed and recreated from
- * one song to the next.
- *
- * <b>STEP 3: Specify a MidiDriver to send events to.</b>
- * MidiParser works by sending MIDI and meta events to a
- * MidiDriver. In the simplest configuration, you can plug
- * a single MidiParser directly into the output MidiDriver
- * being used. However, you can only plug in one at a time;
- * otherwise channel conflicts will occur. Furthermore,
- * meta events that may be needed to interactively control
- * music flow cannot be handled because they are being
- * sent directly to the output device.
- *
- * If you need more control over the MidiParser while it's
- * playing, you can create your own "pseudo-MidiDriver" and
- * place it in between your MidiParser and the output
- * MidiDriver. The MidiParser will send events to your
- * pseudo-MidiDriver, which in turn must send them to the
- * output MidiDriver (or do whatever special handling is
- * required).
- *
- * To specify the MidiDriver to send music output to,
- * use the MidiParser::setMidiDriver method.
- *
- * <b>STEP 4: Specify the onTimer call rate.</b>
- * MidiParser bases the timing of its parsing on an external
- * clock. Every time MidiParser::onTimer is called, a bit
- * more music is parsed. You must specify how many
- * microseconds will occur between each call to onTimer,
- * in order to ensure an accurate music tempo.
- *
- * To set the onTimer call rate, in microseconds,
- * use the MidiParser::setTimerRate method. The onTimer
- * call rate will typically match the timer rate for
- * the output MidiDriver used. This rate can be obtained
- * by calling MidiDriver::getBaseTempo.
- *
- * <b>STEP 5: Load the music.</b>
- * MidiParser requires that the music data already be loaded
- * into memory. The client code is responsible for memory
- * management on this block of memory. That means that the
- * client code must ensure that the data remain in memory
- * while the MidiParser is using it, and properly freed
- * after it is no longer needed. Some MidiParser variants may
- * require internal buffers as well; memory management for those
- * buffers is the responsibility of the MidiParser object.
- *
- * To load the music into the MidiParser, use the
- * MidiParser::loadMusic method, specifying a memory pointer
- * to the music data and the size of the data. (NOTE: Some
- * MidiParser variants don't require a size, and 0 is fine.
- * However, when writing client code to use MidiParser, it is
- * best to assume that a valid size will be required.
- *
- * Convention requires that each implementation of
- * MidiParser::loadMusic automatically set up default tempo
- * and current track. This effectively means that the
- * MidiParser will start playing as soon as timer events
- * start coming in.
- *
- * <b>STEP 6: Activate a timer source for the MidiParser.</b>
- * The easiest timer source to use is the timer of the
- * output MidiDriver. You can attach the MidiDriver's
- * timer output directly to a MidiParser by calling
- * MidiDriver::setTimerCallback. In this case, the timer_proc
- * will be the static method MidiParser::timerCallback,
- * and timer_param will be a pointer to your MidiParser object.
- *
- * This configuration only allows one MidiParser to be driven
- * by the MidiDriver at a time. To drive more MidiDrivers, you
- * will need to create a "pseudo-MidiDriver" as described earlier,
- * In such a configuration, the pseudo-MidiDriver should be set
- * as the timer recipient in MidiDriver::setTimerCallback, and
- * could then call MidiParser::onTimer for each MidiParser object.
- *
- * <b>STEP 7: Music shall begin to play!</b>
- * Congratulations! At this point everything should be hooked up
- * and the MidiParser should generate music. Note that there is
- * no way to "stop" the MidiParser. You can "pause" the MidiParser
- * simply by not sending timer events to it, or you can call
- * MidiParser::unloadMusic to permanently stop the music. (This
- * method resets everything and detaches the MidiParser from the
- * memory block containing the music data.)
- */
-class MidiParser {
-protected:
-	uint16    _active_notes[128];   ///< Each uint16 is a bit mask for channels that have that note on.
-	NoteTimer _hanging_notes[32];   ///< Maintains expiration info for up to 32 notes.
-	                                ///< Used for "Smart Jump" and MIDI formats that do not include explicit Note Off events.
-	byte      _hanging_notes_count; ///< Count of hanging notes, used to optimize expiration.
-
-	MidiDriver *_driver;    ///< The device to which all events will be transmitted.
-	uint32 _timer_rate;     ///< The time in microseconds between onTimer() calls. Obtained from the MidiDriver.
-	uint32 _ppqn;           ///< Pulses Per Quarter Note. (We refer to "pulses" as "ticks".)
-	uint32 _tempo;          ///< Microseconds per quarter note.
-	uint32 _psec_per_tick;  ///< Microseconds per tick (_tempo / _ppqn).
-	bool   _autoLoop;       ///< For lightweight clients that don't provide their own flow control.
-	bool   _smartJump;      ///< Support smart expiration of hanging notes when jumping
-	bool   _centerPitchWheelOnUnload;  ///< Center the pitch wheels when unloading a song
-	bool   _sendSustainOffOnNotesOff;   ///< Send a sustain off on a notes off event, stopping hanging notes
-	byte  *_tracks[120];    ///< Multi-track MIDI formats are supported, up to 120 tracks.
-	byte   _num_tracks;     ///< Count of total tracks for multi-track MIDI formats. 1 for single-track formats.
-	byte   _active_track;   ///< Keeps track of the currently active track, in multi-track formats.
-
-	Tracker _position;      ///< The current time/position in the active track.
-	EventInfo _next_event;  ///< The next event to transmit. Events are preparsed
-	                        ///< so each event is parsed only once; this permits
-	                        ///< simulated events in certain formats.
-	bool   _abort_parse;    ///< If a jump or other operation interrupts parsing, flag to abort.
-
-protected:
-	static uint32 readVLQ(byte * &data);
-	virtual void resetTracking();
-	virtual void allNotesOff();
-	virtual void parseNextEvent(EventInfo &info) = 0;
-
-	void activeNote(byte channel, byte note, bool active);
-	void hangingNote(byte channel, byte note, uint32 ticks_left, bool recycle = true);
-	void hangAllActiveNotes();
-
-	virtual void sendToDriver(uint32 b);
-	void sendToDriver(byte status, byte firstOp, byte secondOp) {
-		sendToDriver(status | ((uint32)firstOp << 8) | ((uint32)secondOp << 16));
-	}
-
-	/**
-	 * Platform independent BE uint32 read-and-advance.
-	 * This helper function reads Big Endian 32-bit numbers
-	 * from a memory pointer, at the same time advancing
-	 * the pointer.
-	 */
-	uint32 read4high(byte * &data) {
-		uint32 val = READ_BE_UINT32(data);
-		data += 4;
-		return val;
-	}
-
-	/**
-	 * Platform independent LE uint16 read-and-advance.
-	 * This helper function reads Little Endian 16-bit numbers
-	 * from a memory pointer, at the same time advancing
-	 * the pointer.
-	 */
-	uint16 read2low(byte * &data) {
-		uint16 val = READ_LE_UINT16(data);
-		data += 2;
-		return val;
-	}
-
-public:
-	/**
-	 * Configuration options for MidiParser
-	 * The following options can be set to modify MidiParser's
-	 * behavior.
-	 */
-	enum {
-		/**
-		 * Events containing a pitch bend command should be treated as
-		 * single-byte padding before the  real event. This allows the
-		 * MidiParser to work with some malformed SMF files from Simon 1/2.
-		 */
-		mpMalformedPitchBends = 1,
-
-		/**
-		 * Sets auto-looping, which can be used by lightweight clients
-		 * that don't provide their own flow control.
-		 */
-		mpAutoLoop = 2,
-
-		/**
-		 * Sets smart jumping, which intelligently expires notes that are
-		 * active when a jump is made, rather than just cutting them off.
-		 */
-		mpSmartJump = 3,
-
-		/**
-		 * Center the pitch wheels when unloading music in preparation
-		 * for the next piece of music.
-		 */
-		mpCenterPitchWheelOnUnload = 4,
-
-		/**
-		 * Sends a sustain off event when a notes off event is triggered.
-		 * Stops hanging notes.
-		 */
-		 mpSendSustainOffOnNotesOff = 5
-	};
-
-public:
-	typedef void (*XMidiCallbackProc)(byte eventData, void *refCon);
-
-	MidiParser();
-	virtual ~MidiParser() { allNotesOff(); }
-
-	virtual bool loadMusic(byte *data, uint32 size) = 0;
-	virtual void unloadMusic();
-	virtual void property(int prop, int value);
-
-	void setMidiDriver(MidiDriver *driver) { _driver = driver; }
-	void setTimerRate(uint32 rate) { _timer_rate = rate; }
-	void setTempo(uint32 tempo);
-	void onTimer();
-
-	bool isPlaying() const { return (_position._play_pos != 0); }
-	void stopPlaying();
-
-	bool setTrack(int track);
-	bool jumpToTick(uint32 tick, bool fireEvents = false, bool stopNotes = true, bool dontSendNoteOn = false);
-
-	uint32 getPPQN() { return _ppqn; }
-	virtual uint32 getTick() { return _position._play_tick; }
-
-	static void defaultXMidiCallback(byte eventData, void *refCon);
-
-	static MidiParser *createParser_SMF();
-	static MidiParser *createParser_XMIDI(XMidiCallbackProc proc = defaultXMidiCallback, void *refCon = 0);
-	static void timerCallback(void *data) { ((MidiParser *) data)->onTimer(); }
-};
-
-#endif
diff --git a/sound/midiparser_smf.cpp b/sound/midiparser_smf.cpp
deleted file mode 100644
index a9c6f1eb3b..0000000000
--- a/sound/midiparser_smf.cpp
+++ /dev/null
@@ -1,384 +0,0 @@
-/* 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 "sound/midiparser.h"
-#include "sound/mididrv.h"
-#include "common/util.h"
-
-/**
- * The Standard MIDI File version of MidiParser.
- */
-class MidiParser_SMF : public MidiParser {
-protected:
-	byte *_buffer;
-	bool _malformedPitchBends;
-
-protected:
-	void compressToType0();
-	void parseNextEvent(EventInfo &info);
-
-public:
-	MidiParser_SMF() : _buffer(0), _malformedPitchBends(false) {}
-	~MidiParser_SMF();
-
-	bool loadMusic(byte *data, uint32 size);
-	void property(int property, int value);
-};
-
-
-static const byte command_lengths[8] = { 3, 3, 3, 3, 2, 2, 3, 0 };
-static const byte special_lengths[16] = { 0, 2, 3, 2, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 0 };
-
-MidiParser_SMF::~MidiParser_SMF() {
-	free(_buffer);
-}
-
-void MidiParser_SMF::property(int prop, int value) {
-	switch (prop) {
-	case mpMalformedPitchBends:
-		_malformedPitchBends = (value > 0);
-	default:
-		MidiParser::property(prop, value);
-	}
-}
-
-void MidiParser_SMF::parseNextEvent(EventInfo &info) {
-	info.start = _position._play_pos;
-	info.delta = readVLQ(_position._play_pos);
-
-	// Process the next info. If mpMalformedPitchBends
-	// was set, we must skip over any pitch bend events
-	// because they are from Simon games and are not
-	// real pitch bend events, they're just two-byte
-	// prefixes before the real info.
-	do {
-		if ((_position._play_pos[0] & 0xF0) >= 0x80)
-			info.event = *(_position._play_pos++);
-		else
-			info.event = _position._running_status;
-	} while (_malformedPitchBends && (info.event & 0xF0) == 0xE0 && _position._play_pos++);
-	if (info.event < 0x80)
-		return;
-
-	_position._running_status = info.event;
-	switch (info.command()) {
-	case 0x9: // Note On
-		info.basic.param1 = *(_position._play_pos++);
-		info.basic.param2 = *(_position._play_pos++);
-		if (info.basic.param2 == 0)
-			info.event = info.channel() | 0x80;
-		info.length = 0;
-		break;
-
-	case 0xC:
-	case 0xD:
-		info.basic.param1 = *(_position._play_pos++);
-		info.basic.param2 = 0;
-		break;
-
-	case 0x8:
-	case 0xA:
-	case 0xB:
-	case 0xE:
-		info.basic.param1 = *(_position._play_pos++);
-		info.basic.param2 = *(_position._play_pos++);
-		info.length = 0;
-		break;
-
-	case 0xF: // System Common, Meta or SysEx event
-		switch (info.event & 0x0F) {
-		case 0x2: // Song Position Pointer
-			info.basic.param1 = *(_position._play_pos++);
-			info.basic.param2 = *(_position._play_pos++);
-			break;
-
-		case 0x3: // Song Select
-			info.basic.param1 = *(_position._play_pos++);
-			info.basic.param2 = 0;
-			break;
-
-		case 0x6:
-		case 0x8:
-		case 0xA:
-		case 0xB:
-		case 0xC:
-		case 0xE:
-			info.basic.param1 = info.basic.param2 = 0;
-			break;
-
-		case 0x0: // SysEx
-			info.length = readVLQ(_position._play_pos);
-			info.ext.data = _position._play_pos;
-			_position._play_pos += info.length;
-			break;
-
-		case 0xF: // META event
-			info.ext.type = *(_position._play_pos++);
-			info.length = readVLQ(_position._play_pos);
-			info.ext.data = _position._play_pos;
-			_position._play_pos += info.length;
-			break;
-
-		default:
-			warning("MidiParser_SMF::parseNextEvent: Unsupported event code %x", info.event);
-		}
-	}
-}
-
-bool MidiParser_SMF::loadMusic(byte *data, uint32 size) {
-	uint32 len;
-	byte midi_type;
-	uint32 total_size;
-	bool isGMF;
-
-	unloadMusic();
-	byte *pos = data;
-	isGMF = false;
-
-	if (!memcmp(pos, "RIFF", 4)) {
-		// Skip the outer RIFF header.
-		pos += 8;
-	}
-
-	if (!memcmp(pos, "MThd", 4)) {
-		// SMF with MTHd information.
-		pos += 4;
-		len = read4high(pos);
-		if (len != 6) {
-			warning("MThd length 6 expected but found %d", (int)len);
-			return false;
-		}
-
-		// Verify that this MIDI either is a Type 2
-		// or has only 1 track. We do not support
-		// multitrack Type 1 files.
-		_num_tracks = pos[2] << 8 | pos[3];
-		midi_type = pos[1];
-		if (midi_type > 2 /*|| (midi_type < 2 && _num_tracks > 1)*/) {
-			warning("No support for a Type %d MIDI with %d tracks", (int)midi_type, (int)_num_tracks);
-			return false;
-		}
-		_ppqn = pos[4] << 8 | pos[5];
-		pos += len;
-	} else if (!memcmp(pos, "GMF\x1", 4)) {
-		// Older GMD/MUS file with no header info.
-		// Assume 1 track, 192 PPQN, and no MTrk headers.
-		isGMF = true;
-		midi_type = 0;
-		_num_tracks = 1;
-		_ppqn = 192;
-		pos += 7; // 'GMD\x1' + 3 bytes of useless (translate: unknown) information
-	} else {
-		warning("Expected MThd or GMD header but found '%c%c%c%c' instead", pos[0], pos[1], pos[2], pos[3]);
-		return false;
-	}
-
-	// Now we identify and store the location for each track.
-	if (_num_tracks > ARRAYSIZE(_tracks)) {
-		warning("Can only handle %d tracks but was handed %d", (int)ARRAYSIZE(_tracks), (int)_num_tracks);
-		return false;
-	}
-
-	total_size = 0;
-	int tracks_read = 0;
-	while (tracks_read < _num_tracks) {
-		if (memcmp(pos, "MTrk", 4) && !isGMF) {
-			warning("Position: %p ('%c')", pos, *pos);
-			warning("Hit invalid block '%c%c%c%c' while scanning for track locations", pos[0], pos[1], pos[2], pos[3]);
-			return false;
-		}
-
-		// If needed, skip the MTrk and length bytes
-		_tracks[tracks_read] = pos + (isGMF ? 0 : 8);
-		if (!isGMF) {
-			pos += 4;
-			len = read4high(pos);
-			total_size += len;
-			pos += len;
-		} else {
-			// An SMF End of Track meta event must be placed
-			// at the end of the stream.
-			data[size++] = 0xFF;
-			data[size++] = 0x2F;
-			data[size++] = 0x00;
-			data[size++] = 0x00;
-		}
-		++tracks_read;
-	}
-
-	// If this is a Type 1 MIDI, we need to now compress
-	// our tracks down into a single Type 0 track.
-	free(_buffer);
-	_buffer = 0;
-
-	if (midi_type == 1) {
-		// FIXME: Doubled the buffer size to prevent crashes with the
-		// Inherit the Earth MIDIs. Jamieson630 said something about a
-		// better fix, but this will have to do in the meantime.
-		_buffer = (byte *)malloc(size * 2);
-		compressToType0();
-		_num_tracks = 1;
-		_tracks[0] = _buffer;
-	}
-
-	// Note that we assume the original data passed in
-	// will persist beyond this call, i.e. we do NOT
-	// copy the data to our own buffer. Take warning....
-	resetTracking();
-	setTempo(500000);
-	setTrack(0);
-	return true;
-}
-
-void MidiParser_SMF::compressToType0() {
-	// We assume that _buffer has been allocated
-	// to sufficient size for this operation.
-
-	// using 0xFF since it could write track_pos[0 to _num_tracks] here
-	// this would cause some illegal writes and could lead to segfaults
-	// (it crashed for some midis for me, they're not used in any game
-	// scummvm supports though). *Maybe* handle this in another way,
-	// it's at the moment only to be sure, that nothing goes wrong.
-	byte *track_pos[0xFF];
-	byte running_status[0xFF];
-	uint32 track_timer[0xFF];
-	uint32 delta;
-	int i;
-
-	for (i = 0; i < _num_tracks; ++i) {
-		running_status[i] = 0;
-		track_pos[i] = _tracks[i];
-		track_timer[i] = readVLQ(track_pos[i]);
-		running_status[i] = 0;
-	}
-
-	int best_i;
-	uint32 length;
-	byte *output = _buffer;
-	byte *pos, *pos2;
-	byte event;
-	uint32 copy_bytes;
-	bool write;
-	byte active_tracks = (byte)_num_tracks;
-
-	while (active_tracks) {
-		write = true;
-		best_i = 255;
-		for (i = 0; i < _num_tracks; ++i) {
-			if (track_pos[i] && (best_i == 255 || track_timer[i] < track_timer[best_i]))
-				best_i = i;
-		}
-		if (best_i == 255) {
-			warning("Premature end of tracks");
-			break;
-		}
-
-		// Initial VLQ delta computation
-		delta = 0;
-		length = track_timer[best_i];
-		for (i = 0; length; ++i) {
-			delta = (delta << 8) | (length & 0x7F) | (i ? 0x80 : 0);
-			length >>= 7;
-		}
-
-		// Process MIDI event.
-		bool implicitEvent = false;
-		copy_bytes = 0;
-		pos = track_pos[best_i];
-		do {
-			event = *(pos++);
-			if (event < 0x80) {
-				event = running_status[best_i];
-				implicitEvent = true;
-			}
-		} while (_malformedPitchBends && (event & 0xF0) == 0xE0 && pos++);
-		running_status[best_i] = event;
-
-		if (command_lengths[(event >> 4) - 8] > 0) {
-			copy_bytes = command_lengths[(event >> 4) - 8];
-		} else if (special_lengths[(event & 0x0F)] > 0) {
-			copy_bytes = special_lengths[(event & 0x0F)];
-		} else if (event == 0xF0) {
-			// SysEx
-			pos2 = pos;
-			length = readVLQ(pos);
-			copy_bytes = 1 + (pos - pos2) + length;
-		} else if (event == 0xFF) {
-			// META
-			event = *(pos++);
-			if (event == 0x2F && active_tracks > 1) {
-				track_pos[best_i] = 0;
-				write = false;
-			} else {
-				pos2 = pos;
-				length = readVLQ(pos);
-				copy_bytes = 2 + (pos - pos2) + length;
-			}
-			if (event == 0x2F)
-				--active_tracks;
-		} else {
-			warning("Bad MIDI command %02X", (int)event);
-			track_pos[best_i] = 0;
-		}
-
-		// Update all tracks' deltas
-		if (write) {
-			for (i = 0; i < _num_tracks; ++i) {
-				if (track_pos[i] && i != best_i)
-					track_timer[i] -= track_timer[best_i];
-			}
-		}
-
-		if (track_pos[best_i]) {
-			if (write) {
-				track_timer[best_i] = 0;
-
-				// Write VLQ delta
-				while (delta & 0x80) {
-					*output++ = (byte)(delta & 0xFF);
-					delta >>= 8;
-				}
-				*output++ = (byte)(delta & 0xFF);
-
-				// Write MIDI data
-				if (!implicitEvent)
-					++track_pos[best_i];
-				--copy_bytes;
-				*output++ = running_status[best_i];
-				memcpy(output, track_pos[best_i], copy_bytes);
-				output += copy_bytes;
-			}
-
-			// Fetch new VLQ delta for winning track
-			track_pos[best_i] += copy_bytes;
-			if (active_tracks)
-				track_timer[best_i] += readVLQ(track_pos[best_i]);
-		}
-	}
-
-	*output++ = 0x00;
-}
-
-MidiParser *MidiParser::createParser_SMF() { return new MidiParser_SMF; }
diff --git a/sound/midiparser_xmidi.cpp b/sound/midiparser_xmidi.cpp
deleted file mode 100644
index 343ca34659..0000000000
--- a/sound/midiparser_xmidi.cpp
+++ /dev/null
@@ -1,375 +0,0 @@
-/* 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 "sound/midiparser.h"
-#include "sound/mididrv.h"
-#include "common/util.h"
-
-/**
- * The XMIDI version of MidiParser.
- *
- * Much of this code is adapted from the XMIDI implementation from the exult
- * project.
- */
-class MidiParser_XMIDI : public MidiParser {
-protected:
-	NoteTimer _notes_cache[32];
-	uint32 _inserted_delta; // Track simulated deltas for note-off events
-
-	struct Loop {
-		byte *pos;
-		byte repeat;
-	};
-
-	Loop _loop[4];
-	int _loopCount;
-
-	XMidiCallbackProc _callbackProc;
-	void *_callbackData;
-
-protected:
-	uint32 readVLQ2(byte * &data);
-	void resetTracking();
-	void parseNextEvent(EventInfo &info);
-
-public:
-	MidiParser_XMIDI(XMidiCallbackProc proc, void *data) : _inserted_delta(0), _callbackProc(proc), _callbackData(data) {}
-	~MidiParser_XMIDI() { }
-
-	bool loadMusic(byte *data, uint32 size);
-};
-
-
-// This is a special XMIDI variable length quantity
-uint32 MidiParser_XMIDI::readVLQ2(byte * &pos) {
-	uint32 value = 0;
-	while (!(pos[0] & 0x80)) {
-		value += *pos++;
-	}
-	return value;
-}
-
-void MidiParser_XMIDI::parseNextEvent(EventInfo &info) {
-	info.start = _position._play_pos;
-	info.delta = readVLQ2(_position._play_pos) - _inserted_delta;
-
-	// Process the next event.
-	_inserted_delta = 0;
-	info.event = *(_position._play_pos++);
-	switch (info.event >> 4) {
-	case 0x9: // Note On
-		info.basic.param1 = *(_position._play_pos++);
-		info.basic.param2 = *(_position._play_pos++);
-		info.length = readVLQ(_position._play_pos);
-		if (info.basic.param2 == 0) {
-			info.event = info.channel() | 0x80;
-			info.length = 0;
-		}
-		break;
-
-	case 0xC:
-	case 0xD:
-		info.basic.param1 = *(_position._play_pos++);
-		info.basic.param2 = 0;
-		break;
-
-	case 0x8:
-	case 0xA:
-	case 0xE:
-		info.basic.param1 = *(_position._play_pos++);
-		info.basic.param2 = *(_position._play_pos++);
-		break;
-
-	case 0xB:
-		info.basic.param1 = *(_position._play_pos++);
-		info.basic.param2 = *(_position._play_pos++);
-
-		// This isn't a full XMIDI implementation, but it should
-		// hopefully be "good enough" for most things.
-
-		switch (info.basic.param1) {
-		// Simplified XMIDI looping.
-		case 0x74: {	// XMIDI_CONTROLLER_FOR_LOOP
-				byte *pos = _position._play_pos;
-				if (_loopCount < ARRAYSIZE(_loop) - 1)
-					_loopCount++;
-				else
-					warning("XMIDI: Exceeding maximum loop count %d", ARRAYSIZE(_loop));
-
-				_loop[_loopCount].pos = pos;
-				_loop[_loopCount].repeat = info.basic.param2;
-				break;
-			}
-
-		case 0x75:	// XMIDI_CONTORLLER_NEXT_BREAK
-			if (_loopCount >= 0) {
-				if (info.basic.param2 < 64) {
-					// End the current loop.
-					_loopCount--;
-				} else {
-					// Repeat 0 means "loop forever".
-					if (_loop[_loopCount].repeat) {
-						if (--_loop[_loopCount].repeat == 0)
-							_loopCount--;
-						else
-							_position._play_pos = _loop[_loopCount].pos;
-					} else {
-						_position._play_pos = _loop[_loopCount].pos;
-					}
-				}
-			}
-			break;
-
-		case 0x77:	// XMIDI_CONTROLLER_CALLBACK_TRIG
-			if (_callbackProc)
-				_callbackProc(info.basic.param2, _callbackData);
-			break;
-
-		case 0x6e:	// XMIDI_CONTROLLER_CHAN_LOCK
-		case 0x6f:	// XMIDI_CONTROLLER_CHAN_LOCK_PROT
-		case 0x70:	// XMIDI_CONTROLLER_VOICE_PROT
-		case 0x71:	// XMIDI_CONTROLLER_TIMBRE_PROT
-		case 0x72:	// XMIDI_CONTROLLER_BANK_CHANGE
-		case 0x73:	// XMIDI_CONTROLLER_IND_CTRL_PREFIX
-		case 0x76:	// XMIDI_CONTROLLER_CLEAR_BB_COUNT
-		case 0x78:	// XMIDI_CONTROLLER_SEQ_BRANCH_INDEX
-		default:
-			if (info.basic.param1 >= 0x6e && info.basic.param1 <= 0x78) {
-				warning("Unsupported XMIDI controller %d (0x%2x)",
-					info.basic.param1, info.basic.param1);
-			}
-		}
-
-		// Should we really keep passing the XMIDI controller events to
-		// the MIDI driver, or should we turn them into some kind of
-		// NOP events? (Dummy meta events, perhaps?) Ah well, it has
-		// worked so far, so it shouldn't cause any damage...
-
-		break;
-
-	case 0xF: // Meta or SysEx event
-		switch (info.event & 0x0F) {
-		case 0x2: // Song Position Pointer
-			info.basic.param1 = *(_position._play_pos++);
-			info.basic.param2 = *(_position._play_pos++);
-			break;
-
-		case 0x3: // Song Select
-			info.basic.param1 = *(_position._play_pos++);
-			info.basic.param2 = 0;
-			break;
-
-		case 0x6:
-		case 0x8:
-		case 0xA:
-		case 0xB:
-		case 0xC:
-		case 0xE:
-			info.basic.param1 = info.basic.param2 = 0;
-			break;
-
-		case 0x0: // SysEx
-			info.length = readVLQ(_position._play_pos);
-			info.ext.data = _position._play_pos;
-			_position._play_pos += info.length;
-			break;
-
-		case 0xF: // META event
-			info.ext.type = *(_position._play_pos++);
-			info.length = readVLQ(_position._play_pos);
-			info.ext.data = _position._play_pos;
-			_position._play_pos += info.length;
-			if (info.ext.type == 0x51 && info.length == 3) {
-				// Tempo event. We want to make these constant 500,000.
-				info.ext.data[0] = 0x07;
-				info.ext.data[1] = 0xA1;
-				info.ext.data[2] = 0x20;
-			}
-			break;
-
-		default:
-			warning("MidiParser_XMIDI::parseNextEvent: Unsupported event code %x", info.event);
-		}
-	}
-}
-
-bool MidiParser_XMIDI::loadMusic(byte *data, uint32 size) {
-	uint32 i = 0;
-	byte *start;
-	uint32 len;
-	uint32 chunk_len;
-	char buf[32];
-
-	_loopCount = -1;
-
-	unloadMusic();
-	byte *pos = data;
-
-	if (!memcmp(pos, "FORM", 4)) {
-		pos += 4;
-
-		// Read length of
-		len = read4high(pos);
-		start = pos;
-
-		// XDIRless XMIDI, we can handle them here.
-		if (!memcmp(pos, "XMID", 4)) {
-			warning("XMIDI doesn't have XDIR");
-			pos += 4;
-			_num_tracks = 1;
-		} else if (memcmp(pos, "XDIR", 4)) {
-			// Not an XMIDI that we recognise
-			warning("Expected 'XDIR' but found '%c%c%c%c'", pos[0], pos[1], pos[2], pos[3]);
-			return false;
-		} else {
-			// Seems Valid
-			pos += 4;
-			_num_tracks = 0;
-
-			for (i = 4; i < len; i++) {
-				// Read 4 bytes of type
-				memcpy(buf, pos, 4);
-				pos += 4;
-
-				// Read length of chunk
-				chunk_len = read4high(pos);
-
-				// Add eight bytes
-				i += 8;
-
-				if (memcmp(buf, "INFO", 4)) {
-					// Must align
-					pos += (chunk_len + 1) & ~1;
-					i += (chunk_len + 1) & ~1;
-					continue;
-				}
-
-				// Must be at least 2 bytes long
-				if (chunk_len < 2) {
-					warning("Invalid chunk length %d for 'INFO' block", (int)chunk_len);
-					return false;
-				}
-
-				_num_tracks = (byte)read2low(pos);
-
-				if (chunk_len > 2) {
-					warning("Chunk length %d is greater than 2", (int)chunk_len);
-					pos += chunk_len - 2;
-				}
-				break;
-			}
-
-			// Didn't get to fill the header
-			if (_num_tracks == 0) {
-				warning("Didn't find a valid track count");
-				return false;
-			}
-
-			// Ok now to start part 2
-			// Goto the right place
-			pos = start + ((len + 1) & ~1);
-
-			if (memcmp(pos, "CAT ", 4)) {
-				// Not an XMID
-				warning("Expected 'CAT ' but found '%c%c%c%c'", pos[0], pos[1], pos[2], pos[3]);
-				return false;
-			}
-			pos += 4;
-
-			// Now read length of this track
-			len = read4high(pos);
-
-			if (memcmp(pos, "XMID", 4)) {
-				// Not an XMID
-				warning("Expected 'XMID' but found '%c%c%c%c'", pos[0], pos[1], pos[2], pos[3]);
-				return false;
-			}
-			pos += 4;
-
-		}
-
-		// Ok it's an XMIDI.
-		// We're going to identify and store the location for each track.
-		if (_num_tracks > ARRAYSIZE(_tracks)) {
-			warning("Can only handle %d tracks but was handed %d", (int)ARRAYSIZE(_tracks), (int)_num_tracks);
-			return false;
-		}
-
-		int tracks_read = 0;
-		while (tracks_read < _num_tracks) {
-			if (!memcmp(pos, "FORM", 4)) {
-				// Skip this plus the 4 bytes after it.
-				pos += 8;
-			} else if (!memcmp(pos, "XMID", 4)) {
-				// Skip this.
-				pos += 4;
-			} else if (!memcmp(pos, "TIMB", 4)) {
-				// Custom timbres?
-				// We don't support them.
-				// Read the length, skip it, and hope there was nothing there.
-				pos += 4;
-				len = read4high(pos);
-				pos += (len + 1) & ~1;
-			} else if (!memcmp(pos, "EVNT", 4)) {
-				// Ahh! What we're looking for at last.
-				_tracks[tracks_read] = pos + 8; // Skip the EVNT and length bytes
-				pos += 4;
-				len = read4high(pos);
-				pos += (len + 1) & ~1;
-				++tracks_read;
-			} else {
-				warning("Hit invalid block '%c%c%c%c' while scanning for track locations", pos[0], pos[1], pos[2], pos[3]);
-				return false;
-			}
-		}
-
-		// If we got this far, we successfully established
-		// the locations for each of our tracks.
-		// Note that we assume the original data passed in
-		// will persist beyond this call, i.e. we do NOT
-		// copy the data to our own buffer. Take warning....
-		_ppqn = 60;
-		resetTracking();
-		setTempo(500000);
-		_inserted_delta = 0;
-		setTrack(0);
-		return true;
-	}
-
-	return false;
-}
-
-void MidiParser_XMIDI::resetTracking() {
-	MidiParser::resetTracking();
-	_inserted_delta = 0;
-}
-
-void MidiParser::defaultXMidiCallback(byte eventData, void *data) {
-	warning("MidiParser: defaultXMidiCallback(%d)", eventData);
-}
-
-MidiParser *MidiParser::createParser_XMIDI(XMidiCallbackProc proc, void *data) {
-	return new MidiParser_XMIDI(proc, data);
-}
diff --git a/sound/mixer.cpp b/sound/mixer.cpp
deleted file mode 100644
index c40aa95d73..0000000000
--- a/sound/mixer.cpp
+++ /dev/null
@@ -1,556 +0,0 @@
-/* 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/util.h"
-#include "common/system.h"
-
-#include "sound/mixer_intern.h"
-#include "sound/rate.h"
-#include "sound/audiostream.h"
-#include "sound/timestamp.h"
-
-
-namespace Audio {
-
-#pragma mark -
-#pragma mark --- Channel classes ---
-#pragma mark -
-
-
-/**
- * Channel used by the default Mixer implementation.
- */
-class Channel {
-public:
-	Channel(Mixer *mixer, Mixer::SoundType type, AudioStream *stream, DisposeAfterUse::Flag autofreeStream, bool reverseStereo, int id, bool permanent);
-	~Channel();
-
-	/**
-	 * Mixes the channel's samples into the given buffer.
-	 *
-	 * @param data buffer where to mix the data
-	 * @param len  number of sample *pairs*. So a value of
-	 *             10 means that the buffer contains twice 10 sample, each
-	 *             16 bits, for a total of 40 bytes.
-	 */
-	void mix(int16 *data, uint len);
-
-	/**
-	 * Queries whether the channel is still playing or not.
-	 */
-	bool isFinished() const { return _stream->endOfStream(); }
-
-	/**
-	 * Queries whether the channel is a permanent channel.
-	 * A permanent channel is not affected by a Mixer::stopAll
-	 * call.
-	 */
-	bool isPermanent() const { return _permanent; }
-
-	/**
-	 * Returns the id of the channel.
-	 */
-	int getId() const { return _id; }
-
-	/**
-	 * Pauses or unpaused the channel in a recursive fashion.
-	 *
-	 * @param paused true, when the channel should be paused.
-	 *               false when it should be unpaused.
-	 */
-	void pause(bool paused);
-
-	/**
-	 * Queries whether the channel is currently paused.
-	 */
-	bool isPaused() const { return (_pauseLevel != 0); }
-
-	/**
-	 * Sets the channel's own volume.
-	 *
-	 * @param volume new volume
-	 */
-	void setVolume(const byte volume);
-
-	/**
-	 * Sets the channel's balance setting.
-	 *
-	 * @param balance new balance
-	 */
-	void setBalance(const int8 balance);
-
-	/**
-	 * Notifies the channel that the global sound type
-	 * volume settings changed.
-	 */
-	void notifyGlobalVolChange() { updateChannelVolumes(); }
-
-	/**
-	 * Queries how long the channel has been playing.
-	 */
-	Timestamp getElapsedTime();
-
-	/**
-	 * Queries the channel's sound type.
-	 */
-	Mixer::SoundType getType() const { return _type; }
-
-	/**
-	 * Sets the channel's sound handle.
-	 *
-	 * @param handle new handle
-	 */
-	void setHandle(const SoundHandle handle) { _handle = handle; }
-
-	/**
-	 * Queries the channel's sound handle.
-	 */
-	SoundHandle getHandle() const { return _handle; }
-
-private:
-	const Mixer::SoundType _type;
-	SoundHandle _handle;
-	bool _permanent;
-	int _pauseLevel;
-	int _id;
-
-	byte _volume;
-	int8 _balance;
-
-	void updateChannelVolumes();
-	st_volume_t _volL, _volR;
-
-	Mixer *_mixer;
-
-	uint32 _samplesConsumed;
-	uint32 _samplesDecoded;
-	uint32 _mixerTimeStamp;
-	uint32 _pauseStartTime;
-	uint32 _pauseTime;
-
-	DisposeAfterUse::Flag _autofreeStream;
-	RateConverter *_converter;
-	AudioStream *_stream;
-};
-
-#pragma mark -
-#pragma mark --- Mixer ---
-#pragma mark -
-
-
-MixerImpl::MixerImpl(OSystem *system, uint sampleRate)
-	: _syst(system), _sampleRate(sampleRate), _mixerReady(false), _handleSeed(0) {
-
-	assert(sampleRate > 0);
-
-	int i;
-
-	for (i = 0; i < ARRAYSIZE(_volumeForSoundType); i++)
-		_volumeForSoundType[i] = kMaxMixerVolume;
-
-	for (i = 0; i != NUM_CHANNELS; i++)
-		_channels[i] = 0;
-}
-
-MixerImpl::~MixerImpl() {
-	for (int i = 0; i != NUM_CHANNELS; i++)
-		delete _channels[i];
-}
-
-void MixerImpl::setReady(bool ready) {
-	_mixerReady = ready;
-}
-
-uint MixerImpl::getOutputRate() const {
-	return _sampleRate;
-}
-
-void MixerImpl::insertChannel(SoundHandle *handle, Channel *chan) {
-	int index = -1;
-	for (int i = 0; i != NUM_CHANNELS; i++) {
-		if (_channels[i] == 0) {
-			index = i;
-			break;
-		}
-	}
-	if (index == -1) {
-		warning("MixerImpl::out of mixer slots");
-		delete chan;
-		return;
-	}
-
-	_channels[index] = chan;
-
-	SoundHandle chanHandle;
-	chanHandle._val = index + (_handleSeed * NUM_CHANNELS);
-
-	chan->setHandle(chanHandle);
-	_handleSeed++;
-	if (handle)
-		*handle = chanHandle;
-}
-
-void MixerImpl::playStream(
-			SoundType type,
-			SoundHandle *handle,
-			AudioStream *stream,
-			int id, byte volume, int8 balance,
-			DisposeAfterUse::Flag autofreeStream,
-			bool permanent,
-			bool reverseStereo) {
-	Common::StackLock lock(_mutex);
-
-	if (stream == 0) {
-		warning("stream is 0");
-		return;
-	}
-
-
-	assert(_mixerReady);
-
-	// Prevent duplicate sounds
-	if (id != -1) {
-		for (int i = 0; i != NUM_CHANNELS; i++)
-			if (_channels[i] != 0 && _channels[i]->getId() == id) {
-				// Delete the stream if were asked to auto-dispose it.
-				// Note: This could cause trouble if the client code does not
-				// yet expect the stream to be gone. The primary example to
-				// keep in mind here is QueuingAudioStream.
-				// Thus, as a quick rule of thumb, you should never, ever,
-				// try to play QueuingAudioStreams with a sound id.
-				if (autofreeStream == DisposeAfterUse::YES)
-					delete stream;
-				return;
-			}
-	}
-
-#ifdef AUDIO_REVERSE_STEREO
-	reverseStereo = !reverseStereo;
-#endif
-
-	// Create the channel
-	Channel *chan = new Channel(this, type, stream, autofreeStream, reverseStereo, id, permanent);
-	chan->setVolume(volume);
-	chan->setBalance(balance);
-	insertChannel(handle, chan);
-}
-
-void MixerImpl::mixCallback(byte *samples, uint len) {
-	assert(samples);
-
-	Common::StackLock lock(_mutex);
-
-	int16 *buf = (int16 *)samples;
-	len >>= 2;
-
-	// Since the mixer callback has been called, the mixer must be ready...
-	_mixerReady = true;
-
-	//  zero the buf
-	memset(buf, 0, 2 * len * sizeof(int16));
-
-	// mix all channels
-	for (int i = 0; i != NUM_CHANNELS; i++)
-		if (_channels[i]) {
-			if (_channels[i]->isFinished()) {
-				delete _channels[i];
-				_channels[i] = 0;
-			} else if (!_channels[i]->isPaused())
-				_channels[i]->mix(buf, len);
-		}
-}
-
-void MixerImpl::stopAll() {
-	Common::StackLock lock(_mutex);
-	for (int i = 0; i != NUM_CHANNELS; i++) {
-		if (_channels[i] != 0 && !_channels[i]->isPermanent()) {
-			delete _channels[i];
-			_channels[i] = 0;
-		}
-	}
-}
-
-void MixerImpl::stopID(int id) {
-	Common::StackLock lock(_mutex);
-	for (int i = 0; i != NUM_CHANNELS; i++) {
-		if (_channels[i] != 0 && _channels[i]->getId() == id) {
-			delete _channels[i];
-			_channels[i] = 0;
-		}
-	}
-}
-
-void MixerImpl::stopHandle(SoundHandle handle) {
-	Common::StackLock lock(_mutex);
-
-	// Simply ignore stop requests for handles of sounds that already terminated
-	const int index = handle._val % NUM_CHANNELS;
-	if (!_channels[index] || _channels[index]->getHandle()._val != handle._val)
-		return;
-
-	delete _channels[index];
-	_channels[index] = 0;
-}
-
-void MixerImpl::setChannelVolume(SoundHandle handle, byte volume) {
-	Common::StackLock lock(_mutex);
-
-	const int index = handle._val % NUM_CHANNELS;
-	if (!_channels[index] || _channels[index]->getHandle()._val != handle._val)
-		return;
-
-	_channels[index]->setVolume(volume);
-}
-
-void MixerImpl::setChannelBalance(SoundHandle handle, int8 balance) {
-	Common::StackLock lock(_mutex);
-
-	const int index = handle._val % NUM_CHANNELS;
-	if (!_channels[index] || _channels[index]->getHandle()._val != handle._val)
-		return;
-
-	_channels[index]->setBalance(balance);
-}
-
-uint32 MixerImpl::getSoundElapsedTime(SoundHandle handle) {
-	return getElapsedTime(handle).msecs();
-}
-
-Timestamp MixerImpl::getElapsedTime(SoundHandle handle) {
-	Common::StackLock lock(_mutex);
-
-	const int index = handle._val % NUM_CHANNELS;
-	if (!_channels[index] || _channels[index]->getHandle()._val != handle._val)
-		return Timestamp(0, _sampleRate);
-
-	return _channels[index]->getElapsedTime();
-}
-
-void MixerImpl::pauseAll(bool paused) {
-	Common::StackLock lock(_mutex);
-	for (int i = 0; i != NUM_CHANNELS; i++) {
-		if (_channels[i] != 0) {
-			_channels[i]->pause(paused);
-		}
-	}
-}
-
-void MixerImpl::pauseID(int id, bool paused) {
-	Common::StackLock lock(_mutex);
-	for (int i = 0; i != NUM_CHANNELS; i++) {
-		if (_channels[i] != 0 && _channels[i]->getId() == id) {
-			_channels[i]->pause(paused);
-			return;
-		}
-	}
-}
-
-void MixerImpl::pauseHandle(SoundHandle handle, bool paused) {
-	Common::StackLock lock(_mutex);
-
-	// Simply ignore (un)pause requests for sounds that already terminated
-	const int index = handle._val % NUM_CHANNELS;
-	if (!_channels[index] || _channels[index]->getHandle()._val != handle._val)
-		return;
-
-	_channels[index]->pause(paused);
-}
-
-bool MixerImpl::isSoundIDActive(int id) {
-	Common::StackLock lock(_mutex);
-	for (int i = 0; i != NUM_CHANNELS; i++)
-		if (_channels[i] && _channels[i]->getId() == id)
-			return true;
-	return false;
-}
-
-int MixerImpl::getSoundID(SoundHandle handle) {
-	Common::StackLock lock(_mutex);
-	const int index = handle._val % NUM_CHANNELS;
-	if (_channels[index] && _channels[index]->getHandle()._val == handle._val)
-		return _channels[index]->getId();
-	return 0;
-}
-
-bool MixerImpl::isSoundHandleActive(SoundHandle handle) {
-	Common::StackLock lock(_mutex);
-	const int index = handle._val % NUM_CHANNELS;
-	return _channels[index] && _channels[index]->getHandle()._val == handle._val;
-}
-
-bool MixerImpl::hasActiveChannelOfType(SoundType type) {
-	Common::StackLock lock(_mutex);
-	for (int i = 0; i != NUM_CHANNELS; i++)
-		if (_channels[i] && _channels[i]->getType() == type)
-			return true;
-	return false;
-}
-
-void MixerImpl::setVolumeForSoundType(SoundType type, int volume) {
-	assert(0 <= type && type < ARRAYSIZE(_volumeForSoundType));
-
-	// Check range
-	if (volume > kMaxMixerVolume)
-		volume = kMaxMixerVolume;
-	else if (volume < 0)
-		volume = 0;
-
-	// TODO: Maybe we should do logarithmic (not linear) volume
-	// scaling? See also Player_V2::setMasterVolume
-
-	Common::StackLock lock(_mutex);
-	_volumeForSoundType[type] = volume;
-
-	for (int i = 0; i != NUM_CHANNELS; ++i) {
-		if (_channels[i] && _channels[i]->getType() == type)
-			_channels[i]->notifyGlobalVolChange();
-	}
-}
-
-int MixerImpl::getVolumeForSoundType(SoundType type) const {
-	assert(0 <= type && type < ARRAYSIZE(_volumeForSoundType));
-
-	return _volumeForSoundType[type];
-}
-
-
-#pragma mark -
-#pragma mark --- Channel implementations ---
-#pragma mark -
-
-Channel::Channel(Mixer *mixer, Mixer::SoundType type, AudioStream *stream,
-                 DisposeAfterUse::Flag autofreeStream, bool reverseStereo, int id, bool permanent)
-    : _type(type), _mixer(mixer), _id(id), _permanent(permanent), _volume(Mixer::kMaxChannelVolume),
-      _balance(0), _pauseLevel(0), _samplesConsumed(0), _samplesDecoded(0), _mixerTimeStamp(0),
-      _pauseStartTime(0), _pauseTime(0), _autofreeStream(autofreeStream), _converter(0),
-      _stream(stream) {
-	assert(mixer);
-	assert(stream);
-
-	// Get a rate converter instance
-	_converter = makeRateConverter(_stream->getRate(), mixer->getOutputRate(), _stream->isStereo(), reverseStereo);
-}
-
-Channel::~Channel() {
-	delete _converter;
-	if (_autofreeStream == DisposeAfterUse::YES)
-		delete _stream;
-}
-
-void Channel::setVolume(const byte volume) {
-	_volume = volume;
-	updateChannelVolumes();
-}
-
-void Channel::setBalance(const int8 balance) {
-	_balance = balance;
-	updateChannelVolumes();
-}
-
-void Channel::updateChannelVolumes() {
-	// From the channel balance/volume and the global volume, we compute
-	// the effective volume for the left and right channel. Note the
-	// slightly odd divisor: the 255 reflects the fact that the maximal
-	// value for _volume is 255, while the 127 is there because the
-	// balance value ranges from -127 to 127.  The mixer (music/sound)
-	// volume is in the range 0 - kMaxMixerVolume.
-	// Hence, the vol_l/vol_r values will be in that range, too
-
-	int vol = _mixer->getVolumeForSoundType(_type) * _volume;
-
-	if (_balance == 0) {
-		_volL = vol / Mixer::kMaxChannelVolume;
-		_volR = vol / Mixer::kMaxChannelVolume;
-	} else if (_balance < 0) {
-		_volL = vol / Mixer::kMaxChannelVolume;
-		_volR = ((127 + _balance) * vol) / (Mixer::kMaxChannelVolume * 127);
-	} else {
-		_volL = ((127 - _balance) * vol) / (Mixer::kMaxChannelVolume * 127);
-		_volR = vol / Mixer::kMaxChannelVolume;
-	}
-}
-
-void Channel::pause(bool paused) {
-	//assert((paused && _pauseLevel >= 0) || (!paused && _pauseLevel));
-
-	if (paused) {
-		_pauseLevel++;
-
-		if (_pauseLevel == 1)
-			_pauseStartTime = g_system->getMillis();
-	} else if (_pauseLevel > 0) {
-		_pauseLevel--;
-
-		if (!_pauseLevel) {
-			_pauseTime = (g_system->getMillis() - _pauseStartTime);
-			_pauseStartTime = 0;
-		}
-	}
-}
-
-Timestamp Channel::getElapsedTime() {
-	const uint32 rate = _mixer->getOutputRate();
-	uint32 delta = 0;
-
-	Audio::Timestamp ts(0, rate);
-
-	if (_mixerTimeStamp == 0)
-		return ts;
-
-	if (isPaused())
-		delta = _pauseStartTime - _mixerTimeStamp;
-	else
-		delta = g_system->getMillis() - _mixerTimeStamp - _pauseTime;
-
-	// Convert the number of samples into a time duration.
-
-	ts = ts.addFrames(_samplesConsumed);
-	ts = ts.addMsecs(delta);
-
-	// In theory it would seem like a good idea to limit the approximation
-	// so that it never exceeds the theoretical upper bound set by
-	// _samplesDecoded. Meanwhile, back in the real world, doing so makes
-	// the Broken Sword cutscenes noticeably jerkier. I guess the mixer
-	// isn't invoked at the regular intervals that I first imagined.
-
-	return ts;
-}
-
-void Channel::mix(int16 *data, uint len) {
-	assert(_stream);
-
-	if (_stream->endOfData()) {
-		// TODO: call drain method
-	} else {
-		assert(_converter);
-
-		_samplesConsumed = _samplesDecoded;
-		_mixerTimeStamp = g_system->getMillis();
-		_pauseTime = 0;
-		_samplesDecoded += _converter->flow(*_stream, data, len, _volL, _volR);
-	}
-}
-
-} // End of namespace Audio
diff --git a/sound/mixer.h b/sound/mixer.h
deleted file mode 100644
index b2accc550d..0000000000
--- a/sound/mixer.h
+++ /dev/null
@@ -1,265 +0,0 @@
-/* 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_MIXER_H
-#define SOUND_MIXER_H
-
-#include "common/types.h"
-#include "common/mutex.h"
-#include "common/noncopyable.h"
-
-#include "sound/timestamp.h"
-
-class OSystem;
-
-
-namespace Audio {
-
-class AudioStream;
-class Channel;
-class Mixer;
-class MixerImpl;
-
-/**
- * A SoundHandle instances corresponds to a specific sound
- * being played via the mixer. It can be used to control that
- * sound (pause it, stop it, etc.).
- * @see The Mixer class
- */
-class SoundHandle {
-	friend class Channel;
-	friend class MixerImpl;
-	uint32 _val;
-public:
-	inline SoundHandle() : _val(0xFFFFFFFF) {}
-};
-
-/**
- * The main audio mixer handles mixing of an arbitrary number of
- * audio streams (in the form of AudioStream instances).
- */
-class Mixer : Common::NonCopyable {
-public:
-	enum SoundType {
-		kPlainSoundType = 0,
-
-		kMusicSoundType = 1,
-		kSFXSoundType = 2,
-		kSpeechSoundType = 3
-	};
-
-	enum {
-		kMaxChannelVolume = 255,
-		kMaxMixerVolume = 256
-	};
-
-public:
-	Mixer() {}
-	virtual ~Mixer() {}
-
-
-
-	/**
-	 * Is the mixer ready and setup? This may not be the case on systems which
-	 * don't support digital sound output. In that case, the mixer proc may
-	 * never be called. That in turn can cause breakage in games which try to
-	 * sync with an audio stream. In particular, the AdLib MIDI emulation...
-	 *
-	 * @return whether the mixer is ready and setup
-	 *
-	 * @todo get rid of this?
-	 */
-	virtual bool isReady() const = 0;
-
-
-	/**
-	 * Start playing the given audio stream.
-	 *
-	 * Note that the sound id assigned below is unique. At most one stream
-	 * with a given id can play at any given time. Trying to play a sound
-	 * with an id that is already in use causes the new sound to be not played.
-	 *
-	 * @param type	the type (voice/sfx/music) of the stream
-	 * @param handle	a SoundHandle which can be used to reference and control
-	 *                  the stream via suitable mixer methods
-	 * @param stream	the actual AudioStream to be played
-	 * @param id	a unique id assigned to this stream
-	 * @param volume	the volume with which to play the sound, ranging from 0 to 255
-	 * @param balance	the balance with which to play the sound, ranging from -128 to 127
-	 * @param autofreeStream	a flag indicating whether the stream should be
-	 *                          freed after playback finished
-	 * @param permanent	a flag indicating whether a plain stopAll call should
-	 *                  not stop this particular stream
-	 * @param reverseStereo	a flag indicating whether left and right channels shall be swapped
-	 */
-	virtual void playStream(
-		SoundType type,
-		SoundHandle *handle,
-		AudioStream *stream,
-		int id = -1,
-		byte volume = kMaxChannelVolume,
-		int8 balance = 0,
-		DisposeAfterUse::Flag autofreeStream = DisposeAfterUse::YES,
-		bool permanent = false,
-		bool reverseStereo = false) = 0;
-
-	/**
-	 * Stop all currently playing sounds.
-	 */
-	virtual void stopAll() = 0;
-
-	/**
-	 * Stop playing the sound with given ID.
-	 *
-	 * @param id the ID of the sound to affect
-	 */
-	virtual void stopID(int id) = 0;
-
-	/**
-	 * Stop playing the sound corresponding to the given handle.
-	 *
-	 * @param handle the sound to affect
-	 */
-	virtual void stopHandle(SoundHandle handle) = 0;
-
-
-
-	/**
-	 * Pause/unpause all sounds, including all regular and permanent
-	 * channels
-	 *
-	 * @param paused true to pause everything, false to unpause
-	 */
-	virtual void pauseAll(bool paused) = 0;
-
-	/**
-	 * Pause/unpause the sound with the given ID.
-	 *
-	 * @param id the ID of the sound to affect
-	 * @param paused true to pause the sound, false to unpause it
-	 */
-	virtual void pauseID(int id, bool paused) = 0;
-
-	/**
-	 * Pause/unpause the sound corresponding to the given handle.
-	 *
-	 * @param handle the sound to affect
-	 * @param paused true to pause the sound, false to unpause it
-	 */
-	virtual void pauseHandle(SoundHandle handle, bool paused) = 0;
-
-
-
-	/**
-	 * Check if a sound with the given ID is active.
-	 *
-	 * @param id the ID of the sound to query
-	 * @return true if the sound is active
-	 */
-	virtual bool isSoundIDActive(int id) = 0;
-
-	/**
-	 * Get the sound ID of handle sound
-	 *
-	 * @param handle sound to query
-	 * @return sound ID if active
-	 */
-	virtual int getSoundID(SoundHandle handle) = 0;
-
-	/**
-	 * Check if a sound with the given handle is active.
-	 *
-	 * @param handle sound to query
-	 * @return true if the sound is active
-	 */
-	virtual bool isSoundHandleActive(SoundHandle handle) = 0;
-
-
-
-	/**
-	 * Set the channel volume for the given handle.
-	 *
-	 * @param handle the sound to affect
-	 * @param volume the new channel volume (0 - kMaxChannelVolume)
-	 */
-	virtual void setChannelVolume(SoundHandle handle, byte volume) = 0;
-
-	/**
-	 * Set the channel balance for the given handle.
-	 *
-	 * @param handle the sound to affect
-	 * @param balance the new channel balance:
-	 *        (-127 ... 0 ... 127) corresponds to (left ... center ... right)
-	 */
-	virtual void setChannelBalance(SoundHandle handle, int8 balance) = 0;
-
-	/**
-	 * Get approximation of for how long the channel has been playing.
-	 */
-	virtual uint32 getSoundElapsedTime(SoundHandle handle) = 0;
-
-	/**
-	 * Get approximation of for how long the channel has been playing.
-	 */
-	virtual Timestamp getElapsedTime(SoundHandle handle) = 0;
-
-	/**
-	 * Check whether any channel of the given sound type is active.
-	 * For example, this can be used to check whether any SFX sound
-	 * is currently playing, by checking for type kSFXSoundType.
-	 *
-	 * @param  type the sound type to look for
-	 * @return true if any channels of the specified type are active.
-	 */
-	virtual bool hasActiveChannelOfType(SoundType type) = 0;
-
-	/**
-	 * Set the volume for the given sound type.
-	 *
-	 * @param type the sound type
-	 * @param volume the new global volume, 0 - kMaxMixerVolume
-	 */
-	virtual void setVolumeForSoundType(SoundType type, int volume) = 0;
-
-	/**
-	 * Query the global volume.
-	 *
-	 * @param type the sound type
-	 * @return the global music volume, 0 - kMaxMixerVolume
-	 */
-	virtual int getVolumeForSoundType(SoundType type) const = 0;
-
-	/**
-	 * Query the system's audio output sample rate.
-	 *
-	 * @return the output sample rate in Hz
-	 */
-	virtual uint getOutputRate() const = 0;
-};
-
-
-} // End of namespace Audio
-
-#endif
diff --git a/sound/mixer_intern.h b/sound/mixer_intern.h
deleted file mode 100644
index 014be7abf2..0000000000
--- a/sound/mixer_intern.h
+++ /dev/null
@@ -1,135 +0,0 @@
-/* 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_MIXER_INTERN_H
-#define SOUND_MIXER_INTERN_H
-
-#include "common/scummsys.h"
-#include "common/mutex.h"
-#include "sound/mixer.h"
-
-namespace Audio {
-
-/**
- * The (default) implementation of the ScummVM audio mixing subsystem.
- *
- * Backends are responsible for allocating (and later releasing) an instance
- * of this class, which engines can access via OSystem::getMixer().
- *
- * Initialisation of instances of this class usually happens as follows:
- * 1) Creat a new Audio::MixerImpl instance.
- * 2) Set the hardware output sample rate via the setSampleRate() method.
- * 3) Hook up the mixCallback() in a suitable audio processing thread/callback.
- * 4) Change the mixer into ready mode via setReady(true).
- * 5) Start audio processing (e.g. by resuming the audio thread, if applicable).
- *
- * In the future, we might make it possible for backends to provide
- * (partial) alternative implementations of the mixer, e.g. to make
- * better use of native sound mixing support on low-end devices.
- *
- * @see OSystem::getMixer()
- */
-class MixerImpl : public Mixer {
-private:
-	enum {
-		NUM_CHANNELS = 16
-	};
-
-	OSystem *_syst;
-	Common::Mutex _mutex;
-
-	const uint _sampleRate;
-	bool _mixerReady;
-	uint32 _handleSeed;
-
-	int _volumeForSoundType[4];
-	Channel *_channels[NUM_CHANNELS];
-
-
-public:
-
-	MixerImpl(OSystem *system, uint sampleRate);
-	~MixerImpl();
-
-	virtual bool isReady() const { return _mixerReady; }
-
-	virtual void playStream(
-		SoundType type,
-		SoundHandle *handle,
-		AudioStream *input,
-		int id, byte volume, int8 balance,
-		DisposeAfterUse::Flag autofreeStream,
-		bool permanent,
-		bool reverseStereo);
-
-	virtual void stopAll();
-	virtual void stopID(int id);
-	virtual void stopHandle(SoundHandle handle);
-
-	virtual void pauseAll(bool paused);
-	virtual void pauseID(int id, bool paused);
-	virtual void pauseHandle(SoundHandle handle, bool paused);
-
-	virtual bool isSoundIDActive(int id);
-	virtual int getSoundID(SoundHandle handle);
-
-	virtual bool isSoundHandleActive(SoundHandle handle);
-
-	virtual void setChannelVolume(SoundHandle handle, byte volume);
-	virtual void setChannelBalance(SoundHandle handle, int8 balance);
-
-	virtual uint32 getSoundElapsedTime(SoundHandle handle);
-	virtual Timestamp getElapsedTime(SoundHandle handle);
-
-	virtual bool hasActiveChannelOfType(SoundType type);
-
-	virtual void setVolumeForSoundType(SoundType type, int volume);
-	virtual int getVolumeForSoundType(SoundType type) const;
-
-	virtual uint getOutputRate() const;
-
-protected:
-	void insertChannel(SoundHandle *handle, Channel *chan);
-
-public:
-	/**
-	 * The mixer callback function, to be called at regular intervals by
-	 * the backend (e.g. from an audio mixing thread). All the actual mixing
-	 * work is done from here.
-	 */
-	void mixCallback(byte *samples, uint len);
-
-	/**
-	 * Set the internal 'is ready' flag of the mixer.
-	 * Backends should invoke Mixer::setReady(true) once initialisation of
-	 * their audio system has been completed.
-	 */
-	void setReady(bool ready);
-};
-
-
-} // End of namespace Audio
-
-#endif
diff --git a/sound/mods/infogrames.cpp b/sound/mods/infogrames.cpp
deleted file mode 100644
index 048872d4a0..0000000000
--- a/sound/mods/infogrames.cpp
+++ /dev/null
@@ -1,470 +0,0 @@
-/* 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 "sound/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/sound/mods/infogrames.h b/sound/mods/infogrames.h
deleted file mode 100644
index 1f3d74f38b..0000000000
--- a/sound/mods/infogrames.h
+++ /dev/null
@@ -1,148 +0,0 @@
-/* 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 "sound/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/sound/mods/maxtrax.cpp b/sound/mods/maxtrax.cpp
deleted file mode 100644
index 08e73fb96a..0000000000
--- a/sound/mods/maxtrax.cpp
+++ /dev/null
@@ -1,1040 +0,0 @@
-/* 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 "sound/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/sound/mods/maxtrax.h b/sound/mods/maxtrax.h
deleted file mode 100644
index 2c86b70288..0000000000
--- a/sound/mods/maxtrax.h
+++ /dev/null
@@ -1,225 +0,0 @@
-/* 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 "sound/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/sound/mods/module.cpp b/sound/mods/module.cpp
deleted file mode 100644
index 992c2a28e3..0000000000
--- a/sound/mods/module.cpp
+++ /dev/null
@@ -1,252 +0,0 @@
-/* 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 "sound/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/sound/mods/module.h b/sound/mods/module.h
deleted file mode 100644
index 550b63617e..0000000000
--- a/sound/mods/module.h
+++ /dev/null
@@ -1,90 +0,0 @@
-/* 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/sound/mods/paula.cpp b/sound/mods/paula.cpp
deleted file mode 100644
index c39b37f83d..0000000000
--- a/sound/mods/paula.cpp
+++ /dev/null
@@ -1,212 +0,0 @@
-/* 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 "sound/mods/paula.h"
-#include "sound/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/sound/mods/paula.h b/sound/mods/paula.h
deleted file mode 100644
index aa3d5b4ab9..0000000000
--- a/sound/mods/paula.h
+++ /dev/null
@@ -1,210 +0,0 @@
-/* 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 "sound/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/sound/mods/protracker.cpp b/sound/mods/protracker.cpp
deleted file mode 100644
index 797b4c417d..0000000000
--- a/sound/mods/protracker.cpp
+++ /dev/null
@@ -1,466 +0,0 @@
-/* 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 "sound/mods/protracker.h"
-#include "sound/mods/paula.h"
-#include "sound/mods/module.h"
-
-#include "sound/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/sound/mods/protracker.h b/sound/mods/protracker.h
deleted file mode 100644
index af722637c7..0000000000
--- a/sound/mods/protracker.h
+++ /dev/null
@@ -1,57 +0,0 @@
-/* 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/sound/mods/rjp1.cpp b/sound/mods/rjp1.cpp
deleted file mode 100644
index be376d61a4..0000000000
--- a/sound/mods/rjp1.cpp
+++ /dev/null
@@ -1,582 +0,0 @@
-/* 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 "sound/mods/paula.h"
-#include "sound/mods/rjp1.h"
-#include "sound/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/sound/mods/rjp1.h b/sound/mods/rjp1.h
deleted file mode 100644
index e1960921b2..0000000000
--- a/sound/mods/rjp1.h
+++ /dev/null
@@ -1,50 +0,0 @@
-/* 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/sound/mods/soundfx.cpp b/sound/mods/soundfx.cpp
deleted file mode 100644
index 3af8ca19c6..0000000000
--- a/sound/mods/soundfx.cpp
+++ /dev/null
@@ -1,275 +0,0 @@
-/* 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 "sound/mods/paula.h"
-#include "sound/mods/soundfx.h"
-#include "sound/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/sound/mods/soundfx.h b/sound/mods/soundfx.h
deleted file mode 100644
index 089c19d292..0000000000
--- a/sound/mods/soundfx.h
+++ /dev/null
@@ -1,53 +0,0 @@
-/* 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/sound/mods/tfmx.cpp b/sound/mods/tfmx.cpp
deleted file mode 100644
index 6ed1abcfb5..0000000000
--- a/sound/mods/tfmx.cpp
+++ /dev/null
@@ -1,1193 +0,0 @@
-/* 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 "sound/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/sound/mods/tfmx.h b/sound/mods/tfmx.h
deleted file mode 100644
index b24df494cd..0000000000
--- a/sound/mods/tfmx.h
+++ /dev/null
@@ -1,284 +0,0 @@
-/* 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 "sound/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)
diff --git a/sound/module.mk b/sound/module.mk
deleted file mode 100644
index 6cfa165a95..0000000000
--- a/sound/module.mk
+++ /dev/null
@@ -1,61 +0,0 @@
-MODULE := sound
-
-MODULE_OBJS := \
-	audiostream.o \
-	fmopl.o \
-	mididrv.o \
-	midiparser_smf.o \
-	midiparser_xmidi.o \
-	midiparser.o \
-	mixer.o \
-	mpu401.o \
-	musicplugin.o \
-	null.o \
-	timestamp.o \
-	decoders/adpcm.o \
-	decoders/aiff.o \
-	decoders/flac.o \
-	decoders/iff_sound.o \
-	decoders/mac_snd.o \
-	decoders/mp3.o \
-	decoders/raw.o \
-	decoders/vag.o \
-	decoders/voc.o \
-	decoders/vorbis.o \
-	decoders/wave.o \
-	mods/infogrames.o \
-	mods/maxtrax.o \
-	mods/module.o \
-	mods/protracker.o \
-	mods/paula.o \
-	mods/rjp1.o \
-	mods/soundfx.o \
-	mods/tfmx.o \
-	softsynth/adlib.o \
-	softsynth/cms.o \
-	softsynth/opl/dbopl.o \
-	softsynth/opl/dosbox.o \
-	softsynth/opl/mame.o \
-	softsynth/fmtowns_pc98/towns_audio.o \
-	softsynth/fmtowns_pc98/towns_euphony.o \
-	softsynth/fmtowns_pc98/towns_pc98_driver.o \
-	softsynth/fmtowns_pc98/towns_pc98_fmsynth.o \
-	softsynth/appleiigs.o \
-	softsynth/ym2612.o \
-	softsynth/fluidsynth.o \
-	softsynth/mt32.o \
-	softsynth/pcspk.o \
-	softsynth/sid.o \
-	softsynth/wave6581.o
-
-ifndef USE_ARM_SOUND_ASM
-MODULE_OBJS += \
-	rate.o
-else
-MODULE_OBJS += \
-	rate_arm.o \
-	rate_arm_asm.o
-endif
-
-# Include common rules
-include $(srcdir)/rules.mk
diff --git a/sound/mpu401.cpp b/sound/mpu401.cpp
deleted file mode 100644
index fe18a3ee85..0000000000
--- a/sound/mpu401.cpp
+++ /dev/null
@@ -1,145 +0,0 @@
-/* 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 "sound/mpu401.h"
-#include "common/system.h"
-#include "common/timer.h"
-#include "common/util.h"	// for ARRAYSIZE
-
-void MidiChannel_MPU401::init(MidiDriver *owner, byte channel) {
-	_owner = owner;
-	_channel = channel;
-	_allocated = false;
-}
-
-bool MidiChannel_MPU401::allocate() {
-	if (_allocated)
-		return false;
-	return (_allocated = true);
-}
-
-MidiDriver *MidiChannel_MPU401::device() {
-	return _owner;
-}
-
-void MidiChannel_MPU401::send(uint32 b) {
-	_owner->send((b & 0xFFFFFFF0) | (_channel & 0xF));
-}
-
-void MidiChannel_MPU401::noteOff(byte note) {
-	_owner->send(note << 8 | 0x80 | _channel);
-}
-
-void MidiChannel_MPU401::noteOn(byte note, byte velocity) {
-	_owner->send(velocity << 16 | note << 8 | 0x90 | _channel);
-}
-
-void MidiChannel_MPU401::programChange(byte program) {
-	_owner->send(program << 8 | 0xC0 | _channel);
-}
-
-void MidiChannel_MPU401::pitchBend(int16 bend) {
-	_owner->send((((bend + 0x2000) >> 7) & 0x7F) << 16 | ((bend + 0x2000) & 0x7F) << 8 | 0xE0 | _channel);
-}
-
-void MidiChannel_MPU401::controlChange(byte control, byte value) {
-	_owner->send(value << 16 | control << 8 | 0xB0 | _channel);
-}
-
-void MidiChannel_MPU401::pitchBendFactor(byte value) {
-	_owner->setPitchBendRange(_channel, value);
-}
-
-void MidiChannel_MPU401::sysEx_customInstrument(uint32 type, const byte *instr) {
-	_owner->sysEx_customInstrument(_channel, type, instr);
-}
-
-const char *MidiDriver::getErrorName(int error_code) {
-	static const char *const midi_errors[] = {
-		"No error",
-		"Cannot connect",
-		"Streaming not available",
-		"Device not available",
-		"Driver already open"
-	};
-
-	if ((uint)error_code >= ARRAYSIZE(midi_errors))
-		return "Unknown Error";
-	return midi_errors[error_code];
-}
-
-MidiDriver_MPU401::MidiDriver_MPU401() :
-	MidiDriver(),
-	_timer_proc(0),
-	_channel_mask(0xFFFF) // Permit all 16 channels by default
-{
-
-	uint i;
-	for (i = 0; i < ARRAYSIZE(_midi_channels); ++i) {
-		_midi_channels[i].init(this, i);
-	}
-}
-
-void MidiDriver_MPU401::close() {
-	if (_timer_proc)
-		g_system->getTimerManager()->removeTimerProc(_timer_proc);
-	_timer_proc = 0;
-	for (int i = 0; i < 16; ++i)
-		send(0x7B << 8 | 0xB0 | i);
-}
-
-uint32 MidiDriver_MPU401::property(int prop, uint32 param) {
-	switch (prop) {
-	case PROP_CHANNEL_MASK:
-		_channel_mask = param & 0xFFFF;
-		return 1;
-	}
-
-	return 0;
-}
-
-MidiChannel *MidiDriver_MPU401::allocateChannel() {
-	MidiChannel_MPU401 *chan;
-	uint i;
-
-	for (i = 0; i < ARRAYSIZE(_midi_channels); ++i) {
-		if (i == 9 || !(_channel_mask & (1 << i)))
-			continue;
-		chan = &_midi_channels[i];
-		if (chan->allocate()) {
-			return chan;
-		}
-	}
-	return NULL;
-}
-
-void MidiDriver_MPU401::setTimerCallback(void *timer_param, Common::TimerManager::TimerProc timer_proc) {
-	if (!_timer_proc || !timer_proc) {
-		if (_timer_proc)
-			g_system->getTimerManager()->removeTimerProc(_timer_proc);
-		_timer_proc = timer_proc;
-		if (timer_proc)
-			g_system->getTimerManager()->installTimerProc(timer_proc, 10000, timer_param);
-	}
-}
diff --git a/sound/mpu401.h b/sound/mpu401.h
deleted file mode 100644
index 2253ab429c..0000000000
--- a/sound/mpu401.h
+++ /dev/null
@@ -1,92 +0,0 @@
-/* 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_MPU401_H
-#define SOUND_MPU401_H
-
-#include "sound/mididrv.h"
-
-////////////////////////////////////////
-//
-// Common MPU401 implementation methods
-//
-////////////////////////////////////////
-
-class MidiDriver_MPU401;
-
-class MidiChannel_MPU401 : public MidiChannel {
-
-private:
-	MidiDriver *_owner;
-	bool _allocated;
-	byte _channel;
-
-public:
-	MidiDriver *device();
-	byte getNumber() { return _channel; }
-	void release() { _allocated = false; }
-
-	void send(uint32 b);
-
-	// Regular messages
-	void noteOff(byte note);
-	void noteOn(byte note, byte velocity);
-	void programChange(byte program);
-	void pitchBend(int16 bend);
-
-	// Control Change messages
-	void controlChange(byte control, byte value);
-	void pitchBendFactor(byte value);
-
-	// SysEx messages
-	void sysEx_customInstrument(uint32 type, const byte *instr);
-
-	// Only to be called by the owner
-	void init(MidiDriver *owner, byte channel);
-	bool allocate();
-};
-
-
-
-class MidiDriver_MPU401 : public MidiDriver {
-private:
-	MidiChannel_MPU401 _midi_channels[16];
-	Common::TimerManager::TimerProc _timer_proc;
-	uint16 _channel_mask;
-
-public:
-	MidiDriver_MPU401();
-
-	virtual void close();
-	void setTimerCallback(void *timer_param, Common::TimerManager::TimerProc timer_proc);
-	uint32 getBaseTempo(void) { return 10000; }
-	uint32 property(int prop, uint32 param);
-
-	MidiChannel *allocateChannel();
-	MidiChannel *getPercussionChannel() { return &_midi_channels[9]; }
-};
-
-
-#endif
diff --git a/sound/musicplugin.cpp b/sound/musicplugin.cpp
deleted file mode 100644
index 8078094616..0000000000
--- a/sound/musicplugin.cpp
+++ /dev/null
@@ -1,64 +0,0 @@
-/* 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 "sound/musicplugin.h"
-#include "common/hash-str.h"
-#include "common/translation.h"
-
-MusicDevice::MusicDevice(MusicPluginObject const *musicPlugin, Common::String name, MusicType mt) :
-	_musicDriverName(_(musicPlugin->getName())), _musicDriverId(musicPlugin->getId()),
-	_name(_(name)), _type(mt) {
-}
-
-Common::String MusicDevice::getCompleteName() {
-	Common::String name;
-
-	if (_name.empty()) {
-		// Default device, just show the driver name
-		name = _musicDriverName;
-	} else {
-		// Show both device and driver names
-		name = _name;
-		name += " [";
-		name += _musicDriverName;
-		name += "]";
-	}
-
-	return name;
-}
-
-Common::String MusicDevice::getCompleteId() {
-	Common::String id = _musicDriverId;
-	if (!_name.empty()) {
-		id += "_";
-		id += _name;
-	}
-
-	return id;
-}
-
-MidiDriver::DeviceHandle MusicDevice::getHandle() {
-	return (MidiDriver::DeviceHandle)Common::hashit(getCompleteId().c_str());
-}
diff --git a/sound/musicplugin.h b/sound/musicplugin.h
deleted file mode 100644
index bbb4ed778c..0000000000
--- a/sound/musicplugin.h
+++ /dev/null
@@ -1,125 +0,0 @@
-/* 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_MUSICPLUGIN_H
-#define SOUND_MUSICPLUGIN_H
-
-#include "base/plugins.h"
-#include "sound/mididrv.h"
-#include "common/list.h"
-
-class MusicPluginObject;
-
-/**
- * Description of a Music device. Used to list the devices a Music driver
- * can manage and their capabilities.
- * A device with an empty name means the default device.
- */
-class MusicDevice {
-public:
-	MusicDevice(MusicPluginObject const *musicPlugin, Common::String name, MusicType mt);
-
-	Common::String &getName() { return _name; }
-	Common::String &getMusicDriverName() { return _musicDriverName; }
-	Common::String &getMusicDriverId() { return _musicDriverId; }
-	MusicType getMusicType() { return _type; }
-
-	/**
-	 * Returns a user readable string that contains the name of the current
-	 * device name (if it isn't the default one) and the name of the driver.
-	 */
-	Common::String getCompleteName();
-
-	/**
-	 * Returns a user readable string that contains the name of the current
-	 * device name (if it isn't the default one) and the id of the driver.
-	 */
-	Common::String getCompleteId();
-
-	MidiDriver::DeviceHandle getHandle();
-
-private:
-	Common::String _name;
-	Common::String _musicDriverName;
-	Common::String _musicDriverId;
-	MusicType _type;
-};
-
-/** List of music devices. */
-typedef Common::List<MusicDevice> MusicDevices;
-
-/**
- * A MusicPluginObject is essentially a factory for MidiDriver instances with
- * the added ability of listing the available devices and their capabilities.
- */
-class MusicPluginObject : public PluginObject {
-public:
-	virtual ~MusicPluginObject() {}
-
-	/**
-	 * Returns a unique string identifier which will be used to save the
-	 * selected MIDI driver to the config file.
-	 */
-	virtual const char *getId() const = 0;
-
-	/**
-	 * Returns a list of the available devices.
-	 */
-	virtual MusicDevices getDevices() const = 0;
-
-	/**
-	 * Tries to instantiate a MIDI Driver instance based on the device
-	 * previously detected via MidiDriver::detectDevice()
-	 *
-	 * @param mididriver	Pointer to a pointer which the MusicPluginObject sets
-	 *				to the newly create MidiDriver, or 0 in case of an error
-	 *
-	* @param dev	Pointer to a device to be used then creating the driver instance.
-	 *				Default value of zero for driver types without devices.
-	 *
-	 * @return		a Common::Error describing the error which occurred, or kNoError
-	 */
-	virtual Common::Error createInstance(MidiDriver **mididriver, MidiDriver::DeviceHandle = 0) const = 0;
-};
-
-
-// Music plugins
-
-typedef PluginSubclass<MusicPluginObject> MusicPlugin;
-
-/**
- * Singleton class which manages all Music plugins.
- */
-class MusicManager : public Common::Singleton<MusicManager> {
-private:
-	friend class Common::Singleton<SingletonBaseType>;
-
-public:
-	const MusicPlugin::List &getPlugins() const;
-};
-
-/** Convenience shortcut for accessing the Music manager. */
-#define MusicMan MusicManager::instance()
-
-#endif
diff --git a/sound/null.cpp b/sound/null.cpp
deleted file mode 100644
index 556b96c14c..0000000000
--- a/sound/null.cpp
+++ /dev/null
@@ -1,62 +0,0 @@
-/* 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 "sound/null.h"
-
-Common::Error NullMusicPlugin::createInstance(MidiDriver **mididriver, MidiDriver::DeviceHandle) const {
-	*mididriver = new MidiDriver_NULL();
-
-	return Common::kNoError;
-}
-
-MusicDevices NullMusicPlugin::getDevices() const {
-	MusicDevices devices;
-	devices.push_back(MusicDevice(this, "", MT_NULL));
-	return devices;
-}
-
-class AutoMusicPlugin : public NullMusicPlugin {
-public:
-	const char *getName() const {
-		return _s("<default>");
-	}
-
-	const char *getId() const {
-		return "auto";
-	}
-	MusicDevices getDevices() const;
-};
-
-MusicDevices AutoMusicPlugin::getDevices() const {
-	MusicDevices devices;
-	devices.push_back(MusicDevice(this, "", MT_AUTO));
-	return devices;
-}
-
-//#if PLUGIN_ENABLED_DYNAMIC(NULL)
-	//REGISTER_PLUGIN_DYNAMIC(NULL, PLUGIN_TYPE_MUSIC, NullMusicPlugin);
-//#else
-	REGISTER_PLUGIN_STATIC(AUTO, PLUGIN_TYPE_MUSIC, AutoMusicPlugin);
-	REGISTER_PLUGIN_STATIC(NULL, PLUGIN_TYPE_MUSIC, NullMusicPlugin);
-//#endif
diff --git a/sound/null.h b/sound/null.h
deleted file mode 100644
index d9343701fa..0000000000
--- a/sound/null.h
+++ /dev/null
@@ -1,56 +0,0 @@
-/* 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_NULL_H
-#define SOUND_NULL_H
-
-#include "sound/musicplugin.h"
-#include "sound/mpu401.h"
-#include "common/translation.h"
-
-/* NULL driver */
-class MidiDriver_NULL : public MidiDriver_MPU401 {
-public:
-	int open() { return 0; }
-	void send(uint32 b) { }
-};
-
-
-// Plugin interface
-
-class NullMusicPlugin : public MusicPluginObject {
-public:
-	virtual const char *getName() const {
-		return _s("No music");
-	}
-
-	virtual const char *getId() const {
-		return "null";
-	}
-
-	virtual MusicDevices getDevices() const;
-	Common::Error createInstance(MidiDriver **mididriver, MidiDriver::DeviceHandle = 0) const;
-};
-
-#endif
diff --git a/sound/rate.cpp b/sound/rate.cpp
deleted file mode 100644
index 97521da3e0..0000000000
--- a/sound/rate.cpp
+++ /dev/null
@@ -1,358 +0,0 @@
-/* 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.
- *
- * $URL$
- * $Id$
- *
- */
-
-/*
- * The code in this file is based on code with Copyright 1998 Fabrice Bellard
- * Fabrice original code is part of SoX (http://sox.sourceforge.net).
- * Max Horn adapted that code to the needs of ScummVM and rewrote it partial,
- * in the process removing any use of floating point arithmetic. Various other
- * improvements over the original code were made.
- */
-
-#include "sound/audiostream.h"
-#include "sound/rate.h"
-#include "sound/mixer.h"
-#include "common/frac.h"
-#include "common/util.h"
-
-namespace Audio {
-
-
-/**
- * The size of the intermediate input cache. Bigger values may increase
- * performance, but only until some point (depends largely on cache size,
- * target processor and various other factors), at which it will decrease
- * again.
- */
-#define INTERMEDIATE_BUFFER_SIZE 512
-
-
-/**
- * Audio rate converter based on simple resampling. Used when no
- * interpolation is required.
- *
- * Limited to sampling frequency <= 65535 Hz.
- */
-template<bool stereo, bool reverseStereo>
-class SimpleRateConverter : public RateConverter {
-protected:
-	st_sample_t inBuf[INTERMEDIATE_BUFFER_SIZE];
-	const st_sample_t *inPtr;
-	int inLen;
-
-	/** position of how far output is ahead of input */
-	/** Holds what would have been opos-ipos */
-	long opos;
-
-	/** fractional position increment in the output stream */
-	long opos_inc;
-
-public:
-	SimpleRateConverter(st_rate_t inrate, st_rate_t outrate);
-	int flow(AudioStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r);
-	int drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol) {
-		return ST_SUCCESS;
-	}
-};
-
-
-/*
- * Prepare processing.
- */
-template<bool stereo, bool reverseStereo>
-SimpleRateConverter<stereo, reverseStereo>::SimpleRateConverter(st_rate_t inrate, st_rate_t outrate) {
-	if ((inrate % outrate) != 0) {
-		error("Input rate must be a multiple of output rate to use rate effect");
-	}
-
-	if (inrate >= 65536 || outrate >= 65536) {
-		error("rate effect can only handle rates < 65536");
-	}
-
-	opos = 1;
-
-	/* increment */
-	opos_inc = inrate / outrate;
-
-	inLen = 0;
-}
-
-/*
- * Processed signed long samples from ibuf to obuf.
- * Return number of sample pairs processed.
- */
-template<bool stereo, bool reverseStereo>
-int SimpleRateConverter<stereo, reverseStereo>::flow(AudioStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r) {
-	st_sample_t *ostart, *oend;
-
-	ostart = obuf;
-	oend = obuf + osamp * 2;
-
-	while (obuf < oend) {
-
-		// read enough input samples so that opos >= 0
-		do {
-			// Check if we have to refill the buffer
-			if (inLen == 0) {
-				inPtr = inBuf;
-				inLen = input.readBuffer(inBuf, ARRAYSIZE(inBuf));
-				if (inLen <= 0)
-					return (obuf - ostart) / 2;
-			}
-			inLen -= (stereo ? 2 : 1);
-			opos--;
-			if (opos >= 0) {
-				inPtr += (stereo ? 2 : 1);
-			}
-		} while (opos >= 0);
-
-		st_sample_t out0, out1;
-		out0 = *inPtr++;
-		out1 = (stereo ? *inPtr++ : out0);
-
-		// Increment output position
-		opos += opos_inc;
-
-		// output left channel
-		clampedAdd(obuf[reverseStereo    ], (out0 * (int)vol_l) / Audio::Mixer::kMaxMixerVolume);
-
-		// output right channel
-		clampedAdd(obuf[reverseStereo ^ 1], (out1 * (int)vol_r) / Audio::Mixer::kMaxMixerVolume);
-
-		obuf += 2;
-	}
-	return (obuf - ostart) / 2;
-}
-
-/**
- * Audio rate converter based on simple linear Interpolation.
- *
- * The use of fractional increment allows us to use no buffer. It
- * avoid the problems at the end of the buffer we had with the old
- * method which stored a possibly big buffer of size
- * lcm(in_rate,out_rate).
- *
- * Limited to sampling frequency <= 65535 Hz.
- */
-
-template<bool stereo, bool reverseStereo>
-class LinearRateConverter : public RateConverter {
-protected:
-	st_sample_t inBuf[INTERMEDIATE_BUFFER_SIZE];
-	const st_sample_t *inPtr;
-	int inLen;
-
-	/** fractional position of the output stream in input stream unit */
-	frac_t opos;
-
-	/** fractional position increment in the output stream */
-	frac_t opos_inc;
-
-	/** last sample(s) in the input stream (left/right channel) */
-	st_sample_t ilast0, ilast1;
-	/** current sample(s) in the input stream (left/right channel) */
-	st_sample_t icur0, icur1;
-
-public:
-	LinearRateConverter(st_rate_t inrate, st_rate_t outrate);
-	int flow(AudioStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r);
-	int drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol) {
-		return ST_SUCCESS;
-	}
-};
-
-
-/*
- * Prepare processing.
- */
-template<bool stereo, bool reverseStereo>
-LinearRateConverter<stereo, reverseStereo>::LinearRateConverter(st_rate_t inrate, st_rate_t outrate) {
-	if (inrate >= 65536 || outrate >= 65536) {
-		error("rate effect can only handle rates < 65536");
-	}
-
-	opos = FRAC_ONE;
-
-	// Compute the linear interpolation increment.
-	// This will overflow if inrate >= 2^16, and underflow if outrate >= 2^16.
-	// Also, if the quotient of the two rate becomes too small / too big, that
-	// would cause problems, but since we rarely scale from 1 to 65536 Hz or vice
-	// versa, I think we can live with that limitation ;-).
-	opos_inc = (inrate << FRAC_BITS) / outrate;
-
-	ilast0 = ilast1 = 0;
-	icur0 = icur1 = 0;
-
-	inLen = 0;
-}
-
-/*
- * Processed signed long samples from ibuf to obuf.
- * Return number of sample pairs processed.
- */
-template<bool stereo, bool reverseStereo>
-int LinearRateConverter<stereo, reverseStereo>::flow(AudioStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r) {
-	st_sample_t *ostart, *oend;
-
-	ostart = obuf;
-	oend = obuf + osamp * 2;
-
-	while (obuf < oend) {
-
-		// read enough input samples so that opos < 0
-		while ((frac_t)FRAC_ONE <= opos) {
-			// Check if we have to refill the buffer
-			if (inLen == 0) {
-				inPtr = inBuf;
-				inLen = input.readBuffer(inBuf, ARRAYSIZE(inBuf));
-				if (inLen <= 0)
-					return (obuf - ostart) / 2;
-			}
-			inLen -= (stereo ? 2 : 1);
-			ilast0 = icur0;
-			icur0 = *inPtr++;
-			if (stereo) {
-				ilast1 = icur1;
-				icur1 = *inPtr++;
-			}
-			opos -= FRAC_ONE;
-		}
-
-		// Loop as long as the outpos trails behind, and as long as there is
-		// still space in the output buffer.
-		while (opos < (frac_t)FRAC_ONE && obuf < oend) {
-			// interpolate
-			st_sample_t out0, out1;
-			out0 = (st_sample_t)(ilast0 + (((icur0 - ilast0) * opos + FRAC_HALF) >> FRAC_BITS));
-			out1 = (stereo ?
-						  (st_sample_t)(ilast1 + (((icur1 - ilast1) * opos + FRAC_HALF) >> FRAC_BITS)) :
-						  out0);
-
-			// output left channel
-			clampedAdd(obuf[reverseStereo    ], (out0 * (int)vol_l) / Audio::Mixer::kMaxMixerVolume);
-
-			// output right channel
-			clampedAdd(obuf[reverseStereo ^ 1], (out1 * (int)vol_r) / Audio::Mixer::kMaxMixerVolume);
-
-			obuf += 2;
-
-			// Increment output position
-			opos += opos_inc;
-		}
-	}
-	return (obuf - ostart) / 2;
-}
-
-
-#pragma mark -
-
-
-/**
- * Simple audio rate converter for the case that the inrate equals the outrate.
- */
-template<bool stereo, bool reverseStereo>
-class CopyRateConverter : public RateConverter {
-	st_sample_t *_buffer;
-	st_size_t _bufferSize;
-public:
-	CopyRateConverter() : _buffer(0), _bufferSize(0) {}
-	~CopyRateConverter() {
-		free(_buffer);
-	}
-
-	virtual int flow(AudioStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r) {
-		assert(input.isStereo() == stereo);
-
-		st_sample_t *ptr;
-		st_size_t len;
-
-		st_sample_t *ostart = obuf;
-
-		if (stereo)
-			osamp *= 2;
-
-		// Reallocate temp buffer, if necessary
-		if (osamp > _bufferSize) {
-			free(_buffer);
-			_buffer = (st_sample_t *)malloc(osamp * 2);
-			_bufferSize = osamp;
-		}
-
-		// Read up to 'osamp' samples into our temporary buffer
-		len = input.readBuffer(_buffer, osamp);
-
-		// Mix the data into the output buffer
-		ptr = _buffer;
-		for (; len > 0; len -= (stereo ? 2 : 1)) {
-			st_sample_t out0, out1;
-			out0 = *ptr++;
-			out1 = (stereo ? *ptr++ : out0);
-
-			// output left channel
-			clampedAdd(obuf[reverseStereo    ], (out0 * (int)vol_l) / Audio::Mixer::kMaxMixerVolume);
-
-			// output right channel
-			clampedAdd(obuf[reverseStereo ^ 1], (out1 * (int)vol_r) / Audio::Mixer::kMaxMixerVolume);
-
-			obuf += 2;
-		}
-		return (obuf - ostart) / 2;
-	}
-
-	virtual int drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol) {
-		return ST_SUCCESS;
-	}
-};
-
-
-#pragma mark -
-
-template<bool stereo, bool reverseStereo>
-RateConverter *makeRateConverter(st_rate_t inrate, st_rate_t outrate) {
-	if (inrate != outrate) {
-		if ((inrate % outrate) == 0) {
-			return new SimpleRateConverter<stereo, reverseStereo>(inrate, outrate);
-		} else {
-			return new LinearRateConverter<stereo, reverseStereo>(inrate, outrate);
-		}
-	} else {
-		return new CopyRateConverter<stereo, reverseStereo>();
-	}
-}
-
-/**
- * Create and return a RateConverter object for the specified input and output rates.
- */
-RateConverter *makeRateConverter(st_rate_t inrate, st_rate_t outrate, bool stereo, bool reverseStereo) {
-	if (stereo) {
-		if (reverseStereo)
-			return makeRateConverter<true, true>(inrate, outrate);
-		else
-			return makeRateConverter<true, false>(inrate, outrate);
-	} else
-		return makeRateConverter<false, false>(inrate, outrate);
-}
-
-} // End of namespace Audio
diff --git a/sound/rate.h b/sound/rate.h
deleted file mode 100644
index fb231e4c4a..0000000000
--- a/sound/rate.h
+++ /dev/null
@@ -1,90 +0,0 @@
-/* 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_RATE_H
-#define SOUND_RATE_H
-
-#include "common/scummsys.h"
-#include "engines/engine.h"
-
-class AudioStream;
-
-
-namespace Audio {
-
-typedef int16 st_sample_t;
-typedef uint16 st_volume_t;
-typedef uint32 st_size_t;
-typedef uint32 st_rate_t;
-
-/* Minimum and maximum values a sample can hold. */
-enum {
-	ST_SAMPLE_MAX = 0x7fffL,
-	ST_SAMPLE_MIN = (-ST_SAMPLE_MAX - 1L)
-};
-
-enum {
-	ST_EOF = -1,
-	ST_SUCCESS = 0
-};
-
-static inline void clampedAdd(int16& a, int b) {
-	register int val;
-#ifdef OUTPUT_UNSIGNED_AUDIO
-	val = (a ^ 0x8000) + b;
-#else
-	val = a + b;
-#endif
-
-	if (val > ST_SAMPLE_MAX)
-		val = ST_SAMPLE_MAX;
-	else if (val < ST_SAMPLE_MIN)
-		val = ST_SAMPLE_MIN;
-
-#ifdef OUTPUT_UNSIGNED_AUDIO
-	a = ((int16)val) ^ 0x8000;
-#else
-	a = val;
-#endif
-}
-
-class RateConverter {
-public:
-	RateConverter() {}
-	virtual ~RateConverter() {}
-
-	/**
-	 * @return Number of sample pairs written into the buffer.
-	 */
-	virtual int flow(AudioStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r) = 0;
-
-	virtual int drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol) = 0;
-};
-
-RateConverter *makeRateConverter(st_rate_t inrate, st_rate_t outrate, bool stereo, bool reverseStereo = false);
-
-} // End of namespace Audio
-
-#endif
diff --git a/sound/rate_arm.cpp b/sound/rate_arm.cpp
deleted file mode 100644
index 63008fcb87..0000000000
--- a/sound/rate_arm.cpp
+++ /dev/null
@@ -1,465 +0,0 @@
-/* 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.
- *
- * $URL$
- * $Id$
- *
- */
-
-/*
- * The code in this file, together with the rate_arm_asm.s file offers
- * an ARM optimised version of the code in rate.cpp. The operation of this
- * code should be identical to that of rate.cpp, but faster. The heavy
- * lifting is done in the assembler file.
- *
- * To be as portable as possible we implement the core routines with C
- * linkage in assembly, and implement the C++ routines that call into
- * the C here. The C++ symbol mangling varies wildly between compilers,
- * so this is the simplest way to ensure that the C/C++ combination should
- * work on as many ARM based platforms as possible.
- *
- * Essentially the algorithm herein is the same as that in rate.cpp, so
- * anyone seeking to understand this should attempt to understand that
- * first. That code was based in turn on code with Copyright 1998 Fabrice
- * Bellard - part of SoX (http://sox.sourceforge.net).
- * Max Horn adapted that code to the needs of ScummVM and partially rewrote
- * it, in the process removing any use of floating point arithmetic. Various
- * other improvments over the original code were made.
- */
-
-#include "sound/audiostream.h"
-#include "sound/rate.h"
-#include "sound/mixer.h"
-#include "common/util.h"
-
-//#define DEBUG_RATECONV
-
-namespace Audio {
-
-/**
- * The precision of the fractional computations used by the rate converter.
- * Normally you should never have to modify this value.
- * This stuff is defined in common/frac.h, but we redefine it here as the
- * ARM routine we call doesn't respect those definitions.
- */
-#define FRAC_BITS 16
-#define FRAC_ONE  (1<<FRAC_BITS)
-
-/**
- * The size of the intermediate input cache. Bigger values may increase
- * performance, but only until some point (depends largely on cache size,
- * target processor and various other factors), at which it will decrease
- * again.
- */
-#define INTERMEDIATE_BUFFER_SIZE 512
-
-
-/**
- * Audio rate converter based on simple resampling. Used when no
- * interpolation is required.
- *
- * Limited to sampling frequency <= 65535 Hz.
- */
-typedef struct {
-	const st_sample_t *inPtr;
-	int inLen;
-
-	/** position of how far output is ahead of input */
-	/** Holds what would have been opos-ipos */
-	long opos;
-
-	/** fractional position increment in the output stream */
-	long opos_inc;
-
-	st_sample_t inBuf[INTERMEDIATE_BUFFER_SIZE];
-} SimpleRateDetails;
-
-template<bool stereo, bool reverseStereo>
-class SimpleRateConverter : public RateConverter {
-protected:
-	SimpleRateDetails  sr;
-public:
-	SimpleRateConverter(st_rate_t inrate, st_rate_t outrate);
-	int flow(AudioStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r);
-	int drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol) {
-		return (ST_SUCCESS);
-	}
-};
-
-
-/*
- * Prepare processing.
- */
-template<bool stereo, bool reverseStereo>
-SimpleRateConverter<stereo, reverseStereo>::SimpleRateConverter(st_rate_t inrate, st_rate_t outrate) {
-	if (inrate == outrate) {
-		error("Input and Output rates must be different to use rate effect");
-	}
-
-	if ((inrate % outrate) != 0) {
-		error("Input rate must be a multiple of Output rate to use rate effect");
-	}
-
-	if (inrate >= 65536 || outrate >= 65536) {
-		error("rate effect can only handle rates < 65536");
-	}
-
-	sr.opos = 1;
-
-	/* increment */
-	sr.opos_inc = inrate / outrate;
-
-	sr.inLen = 0;
-}
-
-extern "C" {
-#ifndef IPHONE
-#define ARM_SimpleRate_M _ARM_SimpleRate_M
-#define ARM_SimpleRate_S _ARM_SimpleRate_S
-#define ARM_SimpleRate_R _ARM_SimpleRate_R
-#endif
-}
-
-extern "C" st_sample_t *ARM_SimpleRate_M(
-                                 AudioStream       &input,
-                                 int (*fn)(Audio::AudioStream&,int16*,int),
-                                 SimpleRateDetails *sr,
-                                 st_sample_t       *obuf,
-                                 st_size_t          osamp,
-                                 st_volume_t        vol_l,
-                                 st_volume_t        vol_r);
-
-extern "C" st_sample_t *ARM_SimpleRate_S(
-                                 AudioStream       &input,
-                                 int (*fn)(Audio::AudioStream&,int16*,int),
-                                 SimpleRateDetails *sr,
-                                 st_sample_t       *obuf,
-                                 st_size_t          osamp,
-                                 st_volume_t        vol_l,
-                                 st_volume_t        vol_r);
-
-extern "C" st_sample_t *ARM_SimpleRate_R(
-                                 AudioStream       &input,
-                                 int (*fn)(Audio::AudioStream&,int16*,int),
-                                 SimpleRateDetails *sr,
-                                 st_sample_t       *obuf,
-                                 st_size_t          osamp,
-                                 st_volume_t        vol_l,
-                                 st_volume_t        vol_r);
-
-extern "C" int SimpleRate_readFudge(Audio::AudioStream &input,
-                                    int16 *a, int b)
-{
-#ifdef DEBUG_RATECONV
-	debug("Reading ptr=%x n%d", a, b);
-#endif
-	return input.readBuffer(a, b);
-}
-
-template<bool stereo, bool reverseStereo>
-int SimpleRateConverter<stereo, reverseStereo>::flow(AudioStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r) {
-
-#ifdef DEBUG_RATECONV
-	debug("Simple st=%d rev=%d", stereo, reverseStereo);
-#endif
-	st_sample_t *ostart = obuf;
-
-	if (!stereo) {
-		obuf = ARM_SimpleRate_M(input,
-		                        &SimpleRate_readFudge,
-		                        &sr,
-			                obuf, osamp, vol_l, vol_r);
-	} else if (reverseStereo) {
-		obuf = ARM_SimpleRate_R(input,
-		                        &SimpleRate_readFudge,
-		                        &sr,
-				        obuf, osamp, vol_l, vol_r);
-	} else {
-		obuf = ARM_SimpleRate_S(input,
-		                        &SimpleRate_readFudge,
-		                        &sr,
-				        obuf, osamp, vol_l, vol_r);
-	}
-	return (obuf-ostart)/2;
-}
-
-/**
- * Audio rate converter based on simple linear Interpolation.
- *
- * The use of fractional increment allows us to use no buffer. It
- * avoid the problems at the end of the buffer we had with the old
- * method which stored a possibly big buffer of size
- * lcm(in_rate,out_rate).
- *
- * Limited to sampling frequency <= 65535 Hz.
- */
-
-typedef struct {
-	const st_sample_t *inPtr;
-	int inLen;
-
-	/** position of how far output is ahead of input */
-	/** Holds what would have been opos-ipos<<16 + opos_frac */
-	long opos;
-
-	/** integer position increment in the output stream */
-	long opos_inc;
-
-	/** current sample(s) in the input stream (left/right channel) */
-	st_sample_t icur[2];
-	/** last sample(s) in the input stream (left/right channel) */
-	/** Note, these are deliberately ints, not st_sample_t's */
-	int32 ilast[2];
-
-	st_sample_t inBuf[INTERMEDIATE_BUFFER_SIZE];
-} LinearRateDetails;
-
-extern "C" {
-#ifndef IPHONE
-#define ARM_LinearRate_M _ARM_LinearRate_M
-#define ARM_LinearRate_S _ARM_LinearRate_S
-#define ARM_LinearRate_R _ARM_LinearRate_R
-#endif
-}
-
-extern "C" st_sample_t *ARM_LinearRate_M(
-                                 AudioStream       &input,
-                                 int (*fn)(Audio::AudioStream&,int16*,int),
-                                 LinearRateDetails *lr,
-                                 st_sample_t       *obuf,
-                                 st_size_t          osamp,
-                                 st_volume_t        vol_l,
-                                 st_volume_t        vol_r);
-
-extern "C" st_sample_t *ARM_LinearRate_S(
-                                 AudioStream       &input,
-                                 int (*fn)(Audio::AudioStream&,int16*,int),
-                                 LinearRateDetails *lr,
-                                 st_sample_t       *obuf,
-                                 st_size_t          osamp,
-                                 st_volume_t        vol_l,
-                                 st_volume_t        vol_r);
-
-extern "C" st_sample_t *ARM_LinearRate_R(
-                                 AudioStream       &input,
-                                 int (*fn)(Audio::AudioStream&,int16*,int),
-                                 LinearRateDetails *lr,
-                                 st_sample_t       *obuf,
-                                 st_size_t          osamp,
-                                 st_volume_t        vol_l,
-                                 st_volume_t        vol_r);
-
-template<bool stereo, bool reverseStereo>
-class LinearRateConverter : public RateConverter {
-protected:
-	LinearRateDetails lr;
-
-public:
-	LinearRateConverter(st_rate_t inrate, st_rate_t outrate);
-	int flow(AudioStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r);
-	int drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol) {
-		return (ST_SUCCESS);
-	}
-};
-
-
-/*
- * Prepare processing.
- */
-template<bool stereo, bool reverseStereo>
-LinearRateConverter<stereo, reverseStereo>::LinearRateConverter(st_rate_t inrate, st_rate_t outrate) {
-	unsigned long incr;
-
-	if (inrate == outrate) {
-		error("Input and Output rates must be different to use rate effect");
-	}
-
-	if (inrate >= 65536 || outrate >= 65536) {
-		error("rate effect can only handle rates < 65536");
-	}
-
-	lr.opos = FRAC_ONE;
-
-	/* increment */
-	incr = (inrate << FRAC_BITS) / outrate;
-
-	lr.opos_inc = incr;
-
-	lr.ilast[0] = lr.ilast[1] = 32768;
-	lr.icur[0] = lr.icur[1] = 0;
-
-	lr.inLen = 0;
-}
-
-/*
- * Processed signed long samples from ibuf to obuf.
- * Return number of sample pairs processed.
- */
-template<bool stereo, bool reverseStereo>
-int LinearRateConverter<stereo, reverseStereo>::flow(AudioStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r) {
-
-#ifdef DEBUG_RATECONV
-	debug("Linear st=%d rev=%d", stereo, reverseStereo);
-#endif
-	st_sample_t *ostart = obuf;
-
-	if (!stereo) {
-		obuf = ARM_LinearRate_M(input,
-		                        &SimpleRate_readFudge,
-		                        &lr,
-				        obuf, osamp, vol_l, vol_r);
-	} else if (reverseStereo) {
-		obuf = ARM_LinearRate_R(input,
-		                        &SimpleRate_readFudge,
-		                        &lr,
-				        obuf, osamp, vol_l, vol_r);
-	} else {
-		obuf = ARM_LinearRate_S(input,
-		                        &SimpleRate_readFudge,
-		                        &lr,
-				        obuf, osamp, vol_l, vol_r);
-	}
-	return (obuf-ostart)/2;
-}
-
-
-#pragma mark -
-
-
-/**
- * Simple audio rate converter for the case that the inrate equals the outrate.
- */
-extern "C" {
-#ifndef IPHONE
-#define ARM_CopyRate_M _ARM_CopyRate_M
-#define ARM_CopyRate_S _ARM_CopyRate_S
-#define ARM_CopyRate_R _ARM_CopyRate_R
-#endif
-}
-
-extern "C" st_sample_t *ARM_CopyRate_M(
-                               st_size_t    len,
-                               st_sample_t *obuf,
-                               st_volume_t  vol_l,
-                               st_volume_t  vol_r,
-                               st_sample_t *_buffer);
-
-extern "C" st_sample_t *ARM_CopyRate_S(
-                               st_size_t    len,
-                               st_sample_t *obuf,
-                               st_volume_t  vol_l,
-                               st_volume_t  vol_r,
-                               st_sample_t *_buffer);
-
-extern "C" st_sample_t *ARM_CopyRate_R(
-                               st_size_t    len,
-                               st_sample_t *obuf,
-                               st_volume_t  vol_l,
-                               st_volume_t  vol_r,
-                               st_sample_t *_buffer);
-
-
-template<bool stereo, bool reverseStereo>
-class CopyRateConverter : public RateConverter {
-	st_sample_t *_buffer;
-	st_size_t _bufferSize;
-public:
-	CopyRateConverter() : _buffer(0), _bufferSize(0) {}
-	~CopyRateConverter() {
-		free(_buffer);
-	}
-
-	virtual int flow(AudioStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r) {
-		assert(input.isStereo() == stereo);
-
-#ifdef DEBUG_RATECONV
-		debug("Copy st=%d rev=%d", stereo, reverseStereo);
-#endif
-		st_size_t len;
-                st_sample_t *ostart = obuf;
-
-		if (stereo)
-			osamp *= 2;
-
-		// Reallocate temp buffer, if necessary
-		if (osamp > _bufferSize) {
-			free(_buffer);
-			_buffer = (st_sample_t *)malloc(osamp * 2);
-			_bufferSize = osamp;
-		}
-
-		// Read up to 'osamp' samples into our temporary buffer
-		len = input.readBuffer(_buffer, osamp);
-		if (len <= 0)
-			return 0;
-
-		// Mix the data into the output buffer
-		if (stereo && reverseStereo)
-			obuf = ARM_CopyRate_R(len, obuf, vol_l, vol_r, _buffer);
-		else if (stereo)
-			obuf = ARM_CopyRate_S(len, obuf, vol_l, vol_r, _buffer);
-		else
-			obuf = ARM_CopyRate_M(len, obuf, vol_l, vol_r, _buffer);
-
-		return (obuf-ostart)/2;
-	}
-	virtual int drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol) {
-		return (ST_SUCCESS);
-	}
-};
-
-
-#pragma mark -
-
-
-/**
- * Create and return a RateConverter object for the specified input and output rates.
- */
-RateConverter *makeRateConverter(st_rate_t inrate, st_rate_t outrate, bool stereo, bool reverseStereo) {
-	if (inrate != outrate) {
-		if ((inrate % outrate) == 0) {
-			if (stereo) {
-				if (reverseStereo)
-					return new SimpleRateConverter<true, true>(inrate, outrate);
-				else
-					return new SimpleRateConverter<true, false>(inrate, outrate);
-			} else
-				return new SimpleRateConverter<false, false>(inrate, outrate);
-		} else {
-			if (stereo) {
-				if (reverseStereo)
-					return new LinearRateConverter<true, true>(inrate, outrate);
-				else
-					return new LinearRateConverter<true, false>(inrate, outrate);
-			} else
-				return new LinearRateConverter<false, false>(inrate, outrate);
-		 }
-	} else {
-		if (stereo) {
-			if (reverseStereo)
-				return new CopyRateConverter<true, true>();
-			else
-				return new CopyRateConverter<true, false>();
-		} else
-			return new CopyRateConverter<false, false>();
-	}
-}
-
-} // End of namespace Audio
diff --git a/sound/rate_arm_asm.s b/sound/rate_arm_asm.s
deleted file mode 100644
index 9431ae0649..0000000000
--- a/sound/rate_arm_asm.s
+++ /dev/null
@@ -1,687 +0,0 @@
-@ 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.
-@
-@ $URL$
-@ $Id$
-@
-@ @author Robin Watts (robin@wss.co.uk)
-@
-@ This file, together with rate_arm.cpp, provides an ARM optimised version
-@ of rate.cpp. The algorithm is essentially the same as that within rate.cpp
-@ so to understand this file you should understand rate.cpp first.
-
-        .text
-
-        .global _ARM_CopyRate_M
-        .global _ARM_CopyRate_S
-        .global _ARM_CopyRate_R
-        .global _ARM_SimpleRate_M
-        .global _ARM_SimpleRate_S
-        .global _ARM_SimpleRate_R
-        .global _ARM_LinearRate_M
-        .global _ARM_LinearRate_S
-        .global _ARM_LinearRate_R
-
-_ARM_CopyRate_M:
-        @ r0 = len
-        @ r1 = obuf
-        @ r2 = vol_l
-        @ r3 = vol_r
-        @ <> = ptr
-        LDR     r12,[r13]
-        STMFD   r13!,{r4-r7,r14}
-
-        MOV     r14,#0                  @ r14= 0
-        ORR     r2, r2, r2, LSL #8      @ r2 = vol_l as 16 bits
-        ORR     r3, r3, r3, LSL #8      @ r3 = vol_r as 16 bits
-CopyRate_M_loop:
-        LDRSH   r5, [r12], #2           @ r5 = tmp0 = tmp1 = *ptr++
-        LDRSH   r6, [r1]                @ r6 = obuf[0]
-        LDRSH   r7, [r1, #2]            @ r7 = obuf[1]
-        MUL     r4, r2, r5              @ r4 = tmp0*vol_l
-        MUL     r5, r3, r5              @ r5 = tmp1*vol_r
-
-        ADDS    r6, r4, r6, LSL #16     @ r6 = obuf[0]<<16 + tmp0*vol_l
-        RSCVS   r6, r14,#0x80000000     @ Clamp r6
-        ADDS    r7, r5, r7, LSL #16     @ r7 = obuf[1]<<16 + tmp1*vol_r
-        RSCVS   r7, r14,#0x80000000     @ Clamp r7
-
-        MOV     r6, r6, LSR #16         @ Shift back to halfword
-        MOV     r7, r7, LSR #16         @ Shift back to halfword
-
-        STRH    r6, [r1], #2            @ Store output value
-        STRH    r7, [r1], #2            @ Store output value
-
-        SUBS    r0,r0,#1                @ len--
-        BGT     CopyRate_M_loop         @ and loop
-
-	MOV	r0, r1			@ return obuf
-
-        LDMFD   r13!,{r4-r7,PC}
-
-_ARM_CopyRate_S:
-        @ r0 = len
-        @ r1 = obuf
-        @ r2 = vol_l
-        @ r3 = vol_r
-        @ <> = ptr
-        LDR     r12,[r13]
-        STMFD   r13!,{r4-r7,r14}
-
-        MOV     r14,#0                  @ r14= 0
-        ORR     r2, r2, r2, LSL #8      @ r2 = vol_l as 16 bits
-        ORR     r3, r3, r3, LSL #8      @ r3 = vol_r as 16 bits
-CopyRate_S_loop:
-        LDRSH   r4, [r12],#2            @ r4 = tmp0 = *ptr++
-        LDRSH   r5, [r12],#2            @ r5 = tmp1 = *ptr++
-        LDRSH   r6, [r1]                @ r6 = obuf[0]
-        LDRSH   r7, [r1,#2]             @ r7 = obuf[1]
-        MUL     r4, r2, r4              @ r5 = tmp0*vol_l
-        MUL     r5, r3, r5              @ r6 = tmp1*vol_r
-
-        ADDS    r6, r4, r6, LSL #16     @ r6 = obuf[0]<<16 + tmp0*vol_l
-        RSCVS   r6, r14,#0x80000000     @ Clamp r6
-        ADDS    r7, r5, r7, LSL #16     @ r7 = obuf[1]<<16 + tmp1*vol_r
-        RSCVS   r7, r14,#0x80000000     @ Clamp r7
-
-        MOV     r6, r6, LSR #16         @ Shift back to halfword
-        MOV     r7, r7, LSR #16         @ Shift back to halfword
-
-        STRH    r6, [r1],#2             @ Store output value
-        STRH    r7, [r1],#2             @ Store output value
-
-        SUBS    r0,r0,#2                @ len -= 2
-        BGT     CopyRate_S_loop         @ and loop
-
-	MOV	r0, r1			@ return obuf
-
-        LDMFD   r13!,{r4-r7,PC}
-
-_ARM_CopyRate_R:
-        @ r0 = len
-        @ r1 = obuf
-        @ r2 = vol_l
-        @ r3 = vol_r
-        @ <> = ptr
-        LDR     r12,[r13]
-        STMFD   r13!,{r4-r7,r14}
-
-        MOV     r14,#0                  @ r14= 0
-        ORR     r2, r2, r2, LSL #8      @ r2 = vol_l as 16 bits
-        ORR     r3, r3, r3, LSL #8      @ r3 = vol_r as 16 bits
-CopyRate_R_loop:
-        LDRSH   r5, [r12],#2            @ r5 = tmp1 = *ptr++
-        LDRSH   r4, [r12],#2            @ r4 = tmp0 = *ptr++
-        LDRSH   r6, [r1]                @ r6 = obuf[0]
-        LDRSH   r7, [r1,#2]             @ r7 = obuf[1]
-        MUL     r4, r2, r4              @ r4 = tmp0*vol_l
-        MUL     r5, r3, r5              @ r5 = tmp1*vol_r
-
-        ADDS    r6, r4, r6, LSL #16     @ r6 = obuf[0]<<16 + tmp0*vol_l
-        RSCVS   r6, r14,#0x80000000     @ Clamp r6
-        ADDS    r7, r5, r7, LSL #16     @ r7 = obuf[1]<<16 + tmp1*vol_r
-        RSCVS   r7, r14,#0x80000000     @ Clamp r7
-
-        MOV     r6, r6, LSR #16         @ Shift back to halfword
-        MOV     r7, r7, LSR #16         @ Shift back to halfword
-
-        STRH    r6, [r1],#2             @ Store output value
-        STRH    r7, [r1],#2             @ Store output value
-
-        SUBS    r0,r0,#2                @ len -= 2
-        BGT     CopyRate_R_loop         @ and loop
-
-	MOV	r0, r1			@ return obuf
-
-        LDMFD   r13!,{r4-r7,PC}
-
-_ARM_SimpleRate_M:
-        @ r0 = AudioStream &input
-        @ r1 = input.readBuffer
-        @ r2 = input->sr
-        @ r3 = obuf
-        @ <> = osamp
-        @ <> = vol_l
-        @ <> = vol_r
-        MOV     r12,r13
-        STMFD   r13!,{r0-r2,r4-r8,r10-r11,r14}
-        LDMFD   r12,{r11,r12,r14}       @ r11= osamp
-                                        @ r12= vol_l
-                                        @ r14= vol_r
-        LDMIA   r2,{r0,r1,r2,r8}        @ r0 = inPtr
-                                        @ r1 = inLen
-                                        @ r2 = opos
-                                        @ r8 = opos_inc
-        CMP     r11,#0                  @ if (osamp <= 0)
-        BLE     SimpleRate_M_end        @   bale
-        MOV     r10,#0
-        ORR     r12,r12,r12,LSL #8      @ r12= vol_l as 16 bits
-        ORR     r14,r14,r14,LSL #8      @ r14= vol_r as 16 bits
-SimpleRate_M_loop:
-        SUBS    r1, r1, #1              @ r1 = inLen -= 1
-        BLT     SimpleRate_M_read
-        SUBS    r2, r2, #1              @ r2 = opos--
-        ADDGE   r0, r0, #2              @ if (r2 >= 0) { sr.inPtr++
-        BGE     SimpleRate_M_loop       @                and loop }
-SimpleRate_M_read_return:
-        LDRSH   r5, [r0],#2             @ r5 = tmp1 = *inPtr++
-        LDRSH   r6, [r3]                @ r6 = obuf[0]
-        LDRSH   r7, [r3,#2]             @ r7 = obuf[1]
-        ADD     r2, r2, r8              @ r2 = opos += opos_inc
-        MUL     r4, r12,r5              @ r4 = tmp0*vol_l
-        MUL     r5, r14,r5              @ r5 = tmp1*vol_r
-
-        ADDS    r6, r4, r6, LSL #16     @ r6 = obuf[0]<<16 + tmp0*vol_l
-        RSCVS   r6, r10,#0x80000000     @ Clamp r6
-        ADDS    r7, r5, r7, LSL #16     @ r7 = obuf[1]<<16 + tmp1*vol_r
-        RSCVS   r7, r10,#0x80000000     @ Clamp r7
-
-        MOV     r6, r6, LSR #16         @ Shift back to halfword
-        MOV     r7, r7, LSR #16         @ Shift back to halfword
-
-        STRH    r6, [r3],#2             @ Store output value
-        STRH    r7, [r3],#2             @ Store output value
-
-        SUBS    r11,r11,#1              @ len--
-        BGT     SimpleRate_M_loop       @ and loop
-SimpleRate_M_end:
-        LDR     r14,[r13,#8]            @ r14 = sr
-        ADD     r13,r13,#12             @ Skip over r0-r2 on stack
-        STMIA   r14,{r0,r1,r2}          @ Store back updated values
-
-	MOV	r0, r3			@ return obuf
-
-        LDMFD   r13!,{r4-r8,r10-r11,PC}
-SimpleRate_M_read:
-        LDR     r0, [r13,#8]            @ r0 = sr (8 = 4*2)
-        ADD     r0, r0, #16             @ r0 = inPtr = inBuf
-        STMFD   r13!,{r0,r2-r3,r12,r14}
-
-        MOV     r1, r0                  @ r1 = inBuf
-        LDR     r0, [r13,#20]          @ r0 = AudioStream & input (20 = 4*5)
-        MOV     r2, #512                @ r2 = ARRAYSIZE(inBuf)
-
-        @ Calling back into C++ here. WinCE is fairly easy about such things
-        @ but other OS are more awkward. r9 is preserved for Symbian, and
-        @ we have 3+8+5 = 16 things on the stack (an even number).
-        MOV     r14,PC
-        LDR     PC,[r13,#24]           @ inLen = input.readBuffer(inBuf,512) (24 = 4*6)
-        SUBS    r1, r0, #1              @ r1 = inLen-1
-        LDMFD   r13!,{r0,r2-r3,r12,r14}
-        BLT     SimpleRate_M_end
-        SUBS    r2, r2, #1              @ r2 = opos--
-        ADDGE   r0, r0, #2              @ if (r2 >= 0) { sr.inPtr++
-        BGE     SimpleRate_M_loop       @                and loop }
-        B       SimpleRate_M_read_return
-
-
-_ARM_SimpleRate_S:
-        @ r0 = AudioStream &input
-        @ r1 = input.readBuffer
-        @ r2 = input->sr
-        @ r3 = obuf
-        @ <> = osamp
-        @ <> = vol_l
-        @ <> = vol_r
-        MOV     r12,r13
-        STMFD   r13!,{r0-r2,r4-r8,r10-r11,r14}
-        LDMFD   r12,{r11,r12,r14}       @ r11= osamp
-                                        @ r12= vol_l
-                                        @ r14= vol_r
-        LDMIA   r2,{r0,r1,r2,r8}        @ r0 = inPtr
-                                        @ r1 = inLen
-                                        @ r2 = opos
-                                        @ r8 = opos_inc
-        CMP     r11,#0                  @ if (osamp <= 0)
-        BLE     SimpleRate_S_end        @   bale
-        MOV     r10,#0
-        ORR     r12,r12,r12,LSL #8      @ r12= vol_l as 16 bits
-        ORR     r14,r14,r14,LSL #8      @ r14= vol_r as 16 bits
-SimpleRate_S_loop:
-        SUBS    r1, r1, #2              @ r1 = inLen -= 2
-        BLT     SimpleRate_S_read
-        SUBS    r2, r2, #1              @ r2 = opos--
-        ADDGE   r0, r0, #4              @ if (r2 >= 0) { sr.inPtr += 2
-        BGE     SimpleRate_S_loop       @                and loop }
-SimpleRate_S_read_return:
-        LDRSH   r4, [r0],#2             @ r4 = tmp0 = *inPtr++
-        LDRSH   r5, [r0],#2             @ r5 = tmp1 = *inPtr++
-        LDRSH   r6, [r3]                @ r6 = obuf[0]
-        LDRSH   r7, [r3,#2]             @ r7 = obuf[1]
-        ADD     r2, r2, r8              @ r2 = opos += opos_inc
-        MUL     r4, r12,r4              @ r5 = tmp0*vol_l
-        MUL     r5, r14,r5              @ r6 = tmp1*vol_r
-
-        ADDS    r6, r4, r6, LSL #16     @ r6 = obuf[0]<<16 + tmp0*vol_l
-        RSCVS   r6, r10,#0x80000000     @ Clamp r6
-        ADDS    r7, r5, r7, LSL #16     @ r7 = obuf[1]<<16 + tmp1*vol_r
-        RSCVS   r7, r10,#0x80000000     @ Clamp r7
-
-        MOV     r6, r6, LSR #16         @ Shift back to halfword
-        MOV     r7, r7, LSR #16         @ Shift back to halfword
-
-        STRH    r6, [r3],#2             @ Store output value
-        STRH    r7, [r3],#2             @ Store output value
-
-        SUBS    r11,r11,#1              @ osamp--
-        BGT     SimpleRate_S_loop       @ and loop
-SimpleRate_S_end:
-        LDR     r14,[r13,#8]            @ r14 = sr
-        ADD     r13,r13,#12             @ skip over r0-r2 on stack
-        STMIA   r14,{r0,r1,r2}          @ store back updated values
-	MOV	r0, r3			@ return obuf
-        LDMFD   r13!,{r4-r8,r10-r11,PC}
-SimpleRate_S_read:
-        LDR     r0, [r13,#8]            @ r0 = sr (8 = 4*2)
-        ADD     r0, r0, #16             @ r0 = inPtr = inBuf
-        STMFD   r13!,{r0,r2-r3,r12,r14}
-        MOV     r1, r0                  @ r1 = inBuf
-        LDR     r0, [r13,#20]           @ r0 = AudioStream & input (20 = 4*5)
-        MOV     r2, #512                @ r2 = ARRAYSIZE(inBuf)
-
-        @ Calling back into C++ here. WinCE is fairly easy about such things
-        @ but other OS are more awkward. r9 is preserved for Symbian, and
-        @ we have 3+8+5 = 16 things on the stack (an even number).
-        MOV     r14,PC
-        LDR     PC,[r13,#24]            @ inLen = input.readBuffer(inBuf,512) (24 = 4*6)
-        SUBS    r1, r0, #2              @ r1 = inLen-2
-        LDMFD   r13!,{r0,r2-r3,r12,r14}
-        BLT     SimpleRate_S_end
-        SUBS    r2, r2, #1              @ r2 = opos--
-        ADDGE   r0, r0, #4              @ if (r2 >= 0) { sr.inPtr += 2
-        BGE     SimpleRate_S_loop       @                and loop }
-        B       SimpleRate_S_read_return
-
-
-
-_ARM_SimpleRate_R:
-        @ r0 = AudioStream &input
-        @ r1 = input.readBuffer
-        @ r2 = input->sr
-        @ r3 = obuf
-        @ <> = osamp
-        @ <> = vol_l
-        @ <> = vol_r
-        MOV     r12,r13
-        STMFD   r13!,{r0-r2,r4-r8,r10-r11,r14}
-        LDMFD   r12,{r11,r12,r14}       @ r11= osamp
-                                        @ r12= vol_l
-                                        @ r14= vol_r
-        LDMIA   r2,{r0,r1,r2,r8}        @ r0 = inPtr
-                                        @ r1 = inLen
-                                        @ r2 = opos
-                                        @ r8 = opos_inc
-        CMP     r11,#0                  @ if (osamp <= 0)
-        BLE     SimpleRate_R_end        @   bale
-        MOV     r10,#0
-        ORR     r12,r12,r12,LSL #8      @ r12= vol_l as 16 bits
-        ORR     r14,r14,r14,LSL #8      @ r14= vol_r as 16 bits
-SimpleRate_R_loop:
-        SUBS    r1, r1, #2              @ r1 = inLen -= 2
-        BLT     SimpleRate_R_read
-        SUBS    r2, r2, #1              @ r2 = opos--
-        ADDGE   r0, r0, #4              @ if (r2 >= 0) { sr.inPtr += 2
-        BGE     SimpleRate_R_loop       @                and loop }
-SimpleRate_R_read_return:
-        LDRSH   r5, [r0],#2             @ r5 = tmp0 = *inPtr++
-        LDRSH   r4, [r0],#2             @ r4 = tmp1 = *inPtr++
-        LDRSH   r6, [r3]                @ r6 = obuf[0]
-        LDRSH   r7, [r3,#2]             @ r7 = obuf[1]
-        ADD     r2, r2, r8              @ r2 = opos += opos_inc
-        MUL     r4, r12,r4              @ r5 = tmp0*vol_l
-        MUL     r5, r14,r5              @ r6 = tmp1*vol_r
-
-        ADDS    r6, r4, r6, LSL #16     @ r6 = obuf[0]<<16 + tmp0*vol_l
-        RSCVS   r6, r10,#0x80000000     @ Clamp r6
-        ADDS    r7, r5, r7, LSL #16     @ r7 = obuf[1]<<16 + tmp1*vol_r
-        RSCVS   r7, r10,#0x80000000     @ Clamp r7
-
-        MOV     r6, r6, LSR #16         @ Shift back to halfword
-        MOV     r7, r7, LSR #16         @ Shift back to halfword
-
-        STRH    r6, [r3],#2             @ Store output value
-        STRH    r7, [r3],#2             @ Store output value
-
-        SUBS    r11,r11,#1              @ osamp--
-        BGT     SimpleRate_R_loop       @ and loop
-SimpleRate_R_end:
-        LDR     r14,[r13,#8]            @ r14 = sr
-        ADD     r13,r13,#12             @ Skip over r0-r2 on stack
-        STMIA   r14,{r0,r1,r2}          @ Store back updated values
-	MOV	r0, r3			@ return obuf
-        LDMFD   r13!,{r4-r8,r10-r11,PC}
-SimpleRate_R_read:
-        LDR     r0, [r13,#8]            @ r0 = sr (8 = 4*2)
-        ADD     r0, r0, #16             @ r0 = inPtr = inBuf
-        STMFD   r13!,{r0,r2-r3,r12,r14}
-
-        MOV     r1, r0                  @ r1 = inBuf
-        LDR     r0, [r13,#20]           @ r0 = AudioStream & input (20 = 4*5)
-        MOV     r2, #512                @ r2 = ARRAYSIZE(inBuf)
-
-        @ Calling back into C++ here. WinCE is fairly easy about such things
-        @ but other OS are more awkward. r9 is preserved for Symbian, and
-        @ we have 3+8+5 = 16 things on the stack (an even number).
-        MOV     r14,PC
-        LDR     PC,[r13,#24]            @ inLen = input.readBuffer(inBuf,512) (24 = 4*6)
-        SUBS    r1, r0, #2              @ r1 = inLen-2
-        LDMFD   r13!,{r0,r2-r3,r12,r14}
-        BLT     SimpleRate_R_end
-        SUBS    r2, r2, #1              @ r2 = opos--
-        ADDGE   r0, r0, #4              @ if (r2 >= 0) { sr.inPtr += 2
-        BGE     SimpleRate_R_loop       @                and loop }
-        B       SimpleRate_R_read_return
-
-
-_ARM_LinearRate_M:
-        @ r0 = AudioStream &input
-        @ r1 = input.readBuffer
-        @ r2 = input->sr
-        @ r3 = obuf
-        @ <> = osamp
-        @ <> = vol_l
-        @ <> = vol_r
-        MOV     r12,r13
-        STMFD   r13!,{r0-r1,r4-r11,r14}
-        LDMFD   r12,{r11,r12,r14}       @ r11= osamp
-                                        @ r12= vol_l
-                                        @ r14= vol_r
-        LDMIA   r2,{r0,r1,r8}           @ r0 = inPtr
-                                        @ r1 = inLen
-                                        @ r8 = opos
-        MOV     r10,#0
-        CMP     r11,#0                  @ if (osamp <= 0)
-        BLE     LinearRate_M_end        @   bale
-        ORR     r12,r12,r12,LSL #8      @ r12= vol_l as 16 bits
-        ORR     r14,r14,r14,LSL #8      @ r14= vol_r as 16 bits
-        CMP     r1,#0
-        BGT     LinearRate_M_part2
-
-        @ part1 - read input samples
-LinearRate_M_loop:
-        SUBS    r1, r1, #1              @ r1 = inLen -= 1
-        BLT     LinearRate_M_read
-LinearRate_M_read_return:
-        LDRH    r4, [r2, #16]           @ r4 = icur[0]
-        LDRSH   r5, [r0],#2             @ r5 = tmp1 = *inPtr++
-        SUBS    r8, r8, #65536          @ r8 = opos--
-        STRH    r4, [r2,#22]            @      ilast[0] = icur[0]
-        STRH    r5, [r2,#16]            @      icur[0] = tmp1
-        BGE     LinearRate_M_loop
-
-        @ part2 - form output samples
-LinearRate_M_part2:
-        @ We are guaranteed that opos < 0 here
-        LDR     r6, [r2,#20]            @ r6 = ilast[0]<<16 + 32768
-        LDRSH   r5, [r2,#16]            @ r5 = icur[0]
-        MOV     r4, r8, LSL #16
-        MOV     r4, r4, LSR #16
-        SUB     r5, r5, r6, ASR #16     @ r5 = icur[0] - ilast[0]
-        MLA     r6, r4, r5, r6  @ r6 = (icur[0]-ilast[0])*opos_frac+ilast[0]
-
-        LDRSH   r4, [r3]                @ r4 = obuf[0]
-        LDRSH   r5, [r3,#2]             @ r5 = obuf[1]
-        MOV     r6, r6, ASR #16         @ r6 = tmp0 = tmp1 >>= 16
-        MUL     r7, r12,r6              @ r7 = tmp0*vol_l
-        MUL     r6, r14,r6              @ r6 = tmp1*vol_r
-
-        ADDS    r7, r7, r4, LSL #16     @ r7 = obuf[0]<<16 + tmp0*vol_l
-        RSCVS   r7, r10, #0x80000000    @ Clamp r7
-        ADDS    r6, r6, r5, LSL #16     @ r6 = obuf[1]<<16 + tmp1*vol_r
-        RSCVS   r6, r10, #0x80000000    @ Clamp r6
-
-        MOV     r7, r7, LSR #16         @ Shift back to halfword
-        MOV     r6, r6, LSR #16         @ Shift back to halfword
-
-        LDR     r5, [r2,#12]            @ r5 = opos_inc
-        STRH    r7, [r3],#2             @ Store output value
-        STRH    r6, [r3],#2             @ Store output value
-        SUBS    r11, r11,#1             @ osamp--
-        BLE     LinearRate_M_end        @ end if needed
-
-        ADDS    r8, r8, r5              @ r8 = opos += opos_inc
-        BLT     LinearRate_M_part2
-        B       LinearRate_M_loop
-LinearRate_M_end:
-        ADD     r13,r13,#8
-        STMIA   r2,{r0,r1,r8}
-	MOV	r0, r3			@ return obuf
-        LDMFD   r13!,{r4-r11,PC}
-LinearRate_M_read:
-        ADD     r0, r2, #28             @ r0 = inPtr = inBuf
-        STMFD   r13!,{r0,r2-r3,r12,r14}
-
-        MOV     r1, r0                  @ r1 = inBuf
-        LDR     r0, [r13,#20]           @ r0 = AudioStream & input (20 = 4*5)
-        MOV     r2, #512                @ r2 = ARRAYSIZE(inBuf)
-
-        @ Calling back into C++ here. WinCE is fairly easy about such things
-        @ but other OS are more awkward. r9 is preserved for Symbian, and
-        @ we have 2+9+5 = 16 things on the stack (an even number).
-        MOV     r14,PC
-        LDR     PC,[r13,#24]           @ inLen = input.readBuffer(inBuf,512) (24 = 4*6)
-        SUBS    r1, r0, #1              @ r1 = inLen-1
-        LDMFD   r13!,{r0,r2-r3,r12,r14}
-        BLT     LinearRate_M_end
-        B       LinearRate_M_read_return
-
-_ARM_LinearRate_S:
-        @ r0 = AudioStream &input
-        @ r1 = input.readBuffer
-        @ r2 = input->sr
-        @ r3 = obuf
-        @ <> = osamp
-        @ <> = vol_l
-        @ <> = vol_r
-        MOV     r12,r13
-        STMFD   r13!,{r0-r1,r4-r11,r14}
-        LDMFD   r12,{r11,r12,r14}       @ r11= osamp
-                                        @ r12= vol_l
-                                        @ r14= vol_r
-        LDMIA   r2,{r0,r1,r8}           @ r0 = inPtr
-                                        @ r1 = inLen
-                                        @ r8 = opos
-        CMP     r11,#0                  @ if (osamp <= 0)
-        BLE     LinearRate_S_end        @   bale
-        ORR     r12,r12,r12,LSL #8      @ r12= vol_l as 16 bits
-        ORR     r14,r14,r14,LSL #8      @ r14= vol_r as 16 bits
-        CMP     r1,#0
-        BGT     LinearRate_S_part2
-
-        @ part1 - read input samples
-LinearRate_S_loop:
-        SUBS    r1, r1, #2              @ r1 = inLen -= 2
-        BLT     LinearRate_S_read
-LinearRate_S_read_return:
-        LDR     r10,[r2, #16]           @ r10= icur[0,1]
-        LDRSH   r5, [r0],#2             @ r5 = tmp0 = *inPtr++
-        LDRSH   r6, [r0],#2             @ r5 = tmp1 = *inPtr++
-        SUBS    r8, r8, #65536          @ r8 = opos--
-        STRH    r10,[r2,#22]            @      ilast[0] = icur[0]
-        MOV     r10,r10,LSR #16
-        STRH    r10,[r2,#26]            @      ilast[1] = icur[1]
-        STRH    r5, [r2,#16]            @      icur[0] = tmp0
-        STRH    r6, [r2,#18]            @      icur[1] = tmp1
-        BGE     LinearRate_S_loop
-
-        @ part2 - form output samples
-LinearRate_S_part2:
-        @ We are guaranteed that opos < 0 here
-        LDR     r6, [r2,#20]            @ r6 = ilast[0]<<16 + 32768
-        LDRSH   r5, [r2,#16]            @ r5 = icur[0]
-        MOV     r4, r8, LSL #16
-        MOV     r4, r4, LSR #16
-        SUB     r5, r5, r6, ASR #16     @ r5 = icur[0] - ilast[0]
-        MLA     r6, r4, r5, r6  @ r6 = (icur[0]-ilast[0])*opos_frac+ilast[0]
-
-        LDR     r7, [r2,#24]            @ r7 = ilast[1]<<16 + 32768
-        LDRSH   r5, [r2,#18]            @ r5 = icur[1]
-        LDRSH   r10,[r3]                @ r10= obuf[0]
-        MOV     r6, r6, ASR #16         @ r6 = tmp1 >>= 16
-        SUB     r5, r5, r7, ASR #16     @ r5 = icur[1] - ilast[1]
-        MLA     r7, r4, r5, r7  @ r7 = (icur[1]-ilast[1])*opos_frac+ilast[1]
-
-        LDRSH   r5, [r3,#2]             @ r5 = obuf[1]
-        MOV     r7, r7, ASR #16         @ r7 = tmp0 >>= 16
-        MUL     r7, r12,r7              @ r7 = tmp0*vol_l
-        MUL     r6, r14,r6              @ r6 = tmp1*vol_r
-
-        ADDS    r7, r7, r10, LSL #16    @ r7 = obuf[0]<<16 + tmp0*vol_l
-        MOV     r4, #0
-        RSCVS   r7, r4, #0x80000000     @ Clamp r7
-        ADDS    r6, r6, r5, LSL #16     @ r6 = obuf[1]<<16 + tmp1*vol_r
-        RSCVS   r6, r4, #0x80000000     @ Clamp r6
-
-        MOV     r7, r7, LSR #16         @ Shift back to halfword
-        MOV     r6, r6, LSR #16         @ Shift back to halfword
-
-        LDR     r5, [r2,#12]            @ r5 = opos_inc
-        STRH    r7, [r3],#2             @ Store output value
-        STRH    r6, [r3],#2             @ Store output value
-        SUBS    r11, r11,#1             @ osamp--
-        BLE     LinearRate_S_end        @ and loop
-
-        ADDS    r8, r8, r5              @ r8 = opos += opos_inc
-        BLT     LinearRate_S_part2
-        B       LinearRate_S_loop
-LinearRate_S_end:
-        ADD     r13,r13,#8
-        STMIA   r2,{r0,r1,r8}
-	MOV	r0, r3			@ return obuf
-        LDMFD   r13!,{r4-r11,PC}
-LinearRate_S_read:
-        ADD     r0, r2, #28             @ r0 = inPtr = inBuf
-        STMFD   r13!,{r0,r2-r3,r12,r14}
-
-        MOV     r1, r0                  @ r1 = inBuf
-        LDR     r0, [r13,#20]          @ r0 = AudioStream & input (20 = 4*5)
-        MOV     r2, #512                @ r2 = ARRAYSIZE(inBuf)
-
-        @ Calling back into C++ here. WinCE is fairly easy about such things
-        @ but other OS are more awkward. r9 is preserved for Symbian, and
-        @ we have 2+9+5 = 16 things on the stack (an even number).
-        MOV     r14,PC
-        LDR     PC,[r13,#24]           @ inLen = input.readBuffer(inBuf,512) (24 = 4*6)
-        SUBS    r1, r0, #2              @ r1 = inLen-2
-        LDMFD   r13!,{r0,r2-r3,r12,r14}
-        BLT     LinearRate_S_end
-        B       LinearRate_S_read_return
-
-_ARM_LinearRate_R:
-        @ r0 = AudioStream &input
-        @ r1 = input.readBuffer
-        @ r2 = input->sr
-        @ r3 = obuf
-        @ <> = osamp
-        @ <> = vol_l
-        @ <> = vol_r
-        MOV     r12,r13
-        STMFD   r13!,{r0-r1,r4-r11,r14}
-        LDMFD   r12,{r11,r12,r14}       @ r11= osamp
-                                        @ r12= vol_l
-                                        @ r14= vol_r
-        LDMIA   r2,{r0,r1,r8}           @ r0 = inPtr
-                                        @ r1 = inLen
-                                        @ r8 = opos
-        CMP     r11,#0                  @ if (osamp <= 0)
-        BLE     LinearRate_R_end        @   bale
-        ORR     r12,r12,r12,LSL #8      @ r12= vol_l as 16 bits
-        ORR     r14,r14,r14,LSL #8      @ r14= vol_r as 16 bits
-        CMP     r1,#0
-        BGT     LinearRate_R_part2
-
-        @ part1 - read input samples
-LinearRate_R_loop:
-        SUBS    r1, r1, #2              @ r1 = inLen -= 2
-        BLT     LinearRate_R_read
-LinearRate_R_read_return:
-        LDR     r10,[r2, #16]           @ r10= icur[0,1]
-        LDRSH   r5, [r0],#2             @ r5 = tmp0 = *inPtr++
-        LDRSH   r6, [r0],#2             @ r5 = tmp1 = *inPtr++
-        SUBS    r8, r8, #65536          @ r8 = opos--
-        STRH    r10,[r2,#22]            @ ilast[0] = icur[0]
-        MOV     r10,r10,LSR #16
-        STRH    r10,[r2,#22]            @ ilast[1] = icur[1]
-        STRH    r5, [r2,#16]            @ icur[0] = tmp0
-        STRH    r6, [r2,#18]            @ icur[1] = tmp1
-        BGE     LinearRate_R_loop
-
-        @ part2 - form output samples
-LinearRate_R_part2:
-        @ We are guaranteed that opos < 0 here
-        LDR     r6, [r2,#20]            @ r6 = ilast[0]
-        LDRSH   r5, [r2,#16]            @ r5 = icur[0]
-        MOV     r4, r8, LSL #16
-        MOV     r4, r4, LSR #16
-        SUB     r5, r5, r6, ASR #16     @ r5 = icur[0] - ilast[0]
-        MLA     r6, r4, r5, r6  @ r6 = (icur[0]-ilast[0])*opos_frac+ilast[0]
-
-        LDR     r7, [r2,#24]            @ r7 = ilast[1]
-        LDRSH   r5, [r2,#18]            @ r5 = icur[1]
-        LDR     r10,[r3]                @ r10= obuf[0]
-        MOV     r6, r6, ASR #16         @ r6 = tmp1 >>= 16
-        SUB     r5, r5, r7, ASR #16     @ r5 = icur[1] - ilast[1]
-        MLA     r7, r4, r5, r7  @ r7 = (icur[1]-ilast[1])*opos_frac+ilast[1]
-
-        LDRSH   r5, [r3,#2]             @ r5 = obuf[1]
-        MOV     r7, r7, ASR #16         @ r7 = tmp0 >>= 16
-        MUL     r7, r12,r7              @ r7 = tmp0*vol_l
-        MUL     r6, r14,r6              @ r6 = tmp1*vol_r
-
-        ADDS    r7, r7, r10, LSL #16    @ r7 = obuf[0]<<16 + tmp0*vol_l
-        MOV     r4, #0
-        RSCVS   r7, r4, #0x80000000     @ Clamp r7
-        ADDS    r6, r6, r5, LSL #16     @ r6 = obuf[1]<<16 + tmp1*vol_r
-        RSCVS   r6, r4, #0x80000000     @ Clamp r6
-
-        MOV     r7, r7, LSR #16         @ Shift back to halfword
-        MOV     r6, r6, LSR #16         @ Shift back to halfword
-
-        LDR     r5, [r2,#12]            @ r5 = opos_inc
-        STRH    r6, [r3],#2             @ Store output value
-        STRH    r7, [r3],#2             @ Store output value
-        SUBS    r11, r11,#1             @ osamp--
-        BLE     LinearRate_R_end        @ and loop
-
-        ADDS    r8, r8, r5              @ r8 = opos += opos_inc
-        BLT     LinearRate_R_part2
-        B       LinearRate_R_loop
-LinearRate_R_end:
-        ADD     r13,r13,#8
-        STMIA   r2,{r0,r1,r8}
-	MOV	r0, r3			@ return obuf
-        LDMFD   r13!,{r4-r11,PC}
-LinearRate_R_read:
-        ADD     r0, r2, #28             @ r0 = inPtr = inBuf
-        STMFD   r13!,{r0,r2-r3,r12,r14}
-
-        MOV     r1, r0                  @ r1 = inBuf
-        LDR     r0, [r13,#20]          @ r0 = AudioStream & input (20 = 4*5)
-        MOV     r2, #512                @ r2 = ARRAYSIZE(inBuf)
-
-        @ Calling back into C++ here. WinCE is fairly easy about such things
-        @ but other OS are more awkward. r9 is preserved for Symbian, and
-        @ we have 2+9+5 = 16 things on the stack (an even number).
-        MOV     r14,PC
-        LDR     PC,[r13,#24]           @ inLen = input.readBuffer(inBuf,512) (24 = 4*6)
-        SUBS    r1, r0, #2              @ r1 = inLen-2
-        LDMFD   r13!,{r0,r2-r3,r12,r14}
-        BLT     LinearRate_R_end
-        B       LinearRate_R_read_return
diff --git a/sound/softsynth/adlib.cpp b/sound/softsynth/adlib.cpp
deleted file mode 100644
index ffb359e816..0000000000
--- a/sound/softsynth/adlib.cpp
+++ /dev/null
@@ -1,1617 +0,0 @@
-/* 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 "sound/softsynth/emumidi.h"
-#include "common/debug.h"
-#include "common/util.h"
-#include "sound/fmopl.h"
-#include "sound/musicplugin.h"
-#include "common/translation.h"
-
-#ifdef DEBUG_ADLIB
-static int tick;
-#endif
-
-class MidiDriver_ADLIB;
-struct AdLibVoice;
-
-// We use packing for the following two structs, because the code
-// does simply copy them over from byte streams, without any
-// serialization. Check AdLibPart::sysEx_customInstrument for an
-// example of this.
-//
-// It might be very well possible, that none of the compilers we support
-// add any padding bytes at all, since the structs contain only variables
-// of the type 'byte'. But better safe than sorry.
-#include "common/pack-start.h"
-struct InstrumentExtra {
-	byte a, b, c, d, e, f, g, h;
-} PACKED_STRUCT;
-
-struct AdLibInstrument {
-	byte mod_characteristic;
-	byte mod_scalingOutputLevel;
-	byte mod_attackDecay;
-	byte mod_sustainRelease;
-	byte mod_waveformSelect;
-	byte car_characteristic;
-	byte car_scalingOutputLevel;
-	byte car_attackDecay;
-	byte car_sustainRelease;
-	byte car_waveformSelect;
-	byte feedback;
-	byte flags_a;
-	InstrumentExtra extra_a;
-	byte flags_b;
-	InstrumentExtra extra_b;
-	byte duration;
-
-	AdLibInstrument() { memset(this, 0, sizeof(AdLibInstrument)); }
-} PACKED_STRUCT;
-#include "common/pack-end.h"
-
-class AdLibPart : public MidiChannel {
-	friend class MidiDriver_ADLIB;
-
-protected:
-//	AdLibPart *_prev, *_next;
-	AdLibVoice *_voice;
-	int16 _pitchbend;
-	byte _pitchbend_factor;
-	int8 _transpose_eff;
-	byte _vol_eff;
-	int8 _detune_eff;
-	byte _modwheel;
-	bool _pedal;
-	byte _program;
-	byte _pri_eff;
-	AdLibInstrument _part_instr;
-
-protected:
-	MidiDriver_ADLIB *_owner;
-	bool _allocated;
-	byte _channel;
-
-	void init(MidiDriver_ADLIB *owner, byte channel);
-	void allocate() { _allocated = true; }
-
-public:
-	AdLibPart() {
-		_voice = 0;
-		_pitchbend = 0;
-		_pitchbend_factor = 2;
-		_transpose_eff = 0;
-		_vol_eff = 0;
-		_detune_eff = 0;
-		_modwheel = 0;
-		_pedal = 0;
-		_program = 0;
-		_pri_eff = 0;
-
-		_owner = 0;
-		_allocated = false;
-		_channel = 0;
-	}
-
-	MidiDriver *device();
-	byte getNumber() { return _channel; }
-	void release() { _allocated = false; }
-
-	void send (uint32 b);
-
-	// Regular messages
-	void noteOff(byte note);
-	void noteOn(byte note, byte velocity);
-	void programChange(byte program);
-	void pitchBend(int16 bend);
-
-	// Control Change messages
-	void controlChange(byte control, byte value);
-	void modulationWheel(byte value);
-	void volume(byte value);
-	void panPosition(byte value) { return; } // Not supported
-	void pitchBendFactor(byte value);
-	void detune(byte value);
-	void priority(byte value);
-	void sustain(bool value);
-	void effectLevel(byte value) { return; } // Not supported
-	void chorusLevel(byte value) { return; } // Not supported
-	void allNotesOff();
-
-	// SysEx messages
-	void sysEx_customInstrument(uint32 type, const byte *instr);
-};
-
-// FYI (Jamieson630)
-// It is assumed that any invocation to AdLibPercussionChannel
-// will be done through the MidiChannel base class as opposed to the
-// AdLibPart base class. If this were NOT the case, all the functions
-// listed below would need to be virtual in AdLibPart as well as MidiChannel.
-class AdLibPercussionChannel : public AdLibPart {
-	friend class MidiDriver_ADLIB;
-
-protected:
-	void init(MidiDriver_ADLIB *owner, byte channel);
-
-public:
-	~AdLibPercussionChannel();
-
-	void noteOff(byte note);
-	void noteOn(byte note, byte velocity);
-	void programChange(byte program) { }
-	void pitchBend(int16 bend) { }
-
-	// Control Change messages
-	void modulationWheel(byte value) { }
-	void pitchBendFactor(byte value) { }
-	void detune(byte value) { }
-	void priority(byte value) { }
-	void sustain(bool value) { }
-
-	// SysEx messages
-	void sysEx_customInstrument(uint32 type, const byte *instr);
-
-private:
-	byte _notes[256];
-	AdLibInstrument *_customInstruments[256];
-};
-
-struct Struct10 {
-	byte active;
-	int16 cur_val;
-	int16 count;
-	uint16 max_value;
-	int16 start_value;
-	byte loop;
-	byte table_a[4];
-	byte table_b[4];
-	int8 unk3;
-	int8 modwheel;
-	int8 modwheel_last;
-	uint16 speed_lo_max;
-	uint16 num_steps;
-	int16 speed_hi;
-	int8 direction;
-	uint16 speed_lo;
-	uint16 speed_lo_counter;
-};
-
-struct Struct11 {
-	int16 modify_val;
-	byte param, flag0x40, flag0x10;
-	Struct10 *s10;
-};
-
-struct AdLibVoice {
-	AdLibPart *_part;
-	AdLibVoice *_next, *_prev;
-	byte _waitforpedal;
-	byte _note;
-	byte _channel;
-	byte _twochan;
-	byte _vol_1, _vol_2;
-	int16 _duration;
-
-	Struct10 _s10a;
-	Struct11 _s11a;
-	Struct10 _s10b;
-	Struct11 _s11b;
-
-	AdLibVoice() { memset(this, 0, sizeof(AdLibVoice)); }
-};
-
-struct AdLibSetParams {
-	byte a, b, c, d;
-};
-
-static const byte channel_mappings[9] = {
-	0, 1, 2, 8,
-	9, 10, 16, 17,
-	18
-};
-
-static const byte channel_mappings_2[9] = {
-	3, 4, 5, 11,
-	12, 13, 19, 20,
-	21
-};
-
-static const AdLibSetParams adlib_setparam_table[] = {
-	{0x40, 0, 63, 63},  // level
-	{0xE0, 2, 0, 0},    // unused
-	{0x40, 6, 192, 0},  // level key scaling
-	{0x20, 0, 15, 0},   // modulator frequency multiple
-	{0x60, 4, 240, 15}, // attack rate
-	{0x60, 0, 15, 15},  // decay rate
-	{0x80, 4, 240, 15}, // sustain level
-	{0x80, 0, 15, 15},  // release rate
-	{0xE0, 0, 3, 0},    // waveformSelect select
-	{0x20, 7, 128, 0},  // amp mod
-	{0x20, 6, 64, 0},   // vib
-	{0x20, 5, 32, 0},   // eg typ
-	{0x20, 4, 16, 0},   // ksr
-	{0xC0, 0, 1, 0},    // decay alg
-	{0xC0, 1, 14, 0}    // feedback
-};
-
-static const byte param_table_1[16] = {
-	29, 28, 27, 0,
-	3, 4, 7, 8,
-	13, 16, 17, 20,
-	21, 30, 31, 0
-};
-
-static const uint16 maxval_table[16] = {
-	0x2FF, 0x1F, 0x7, 0x3F,
-	0x0F, 0x0F, 0x0F, 0x3,
-	0x3F, 0x0F, 0x0F, 0x0F,
-	0x3, 0x3E, 0x1F, 0
-};
-
-static const uint16 num_steps_table[] = {
-	1, 2, 4, 5,
-	6, 7, 8, 9,
-	10, 12, 14, 16,
-	18, 21, 24, 30,
-	36, 50, 64, 82,
-	100, 136, 160, 192,
-	240, 276, 340, 460,
-	600, 860, 1200, 1600
-};
-
-static const byte note_to_f_num[] = {
-	90, 91, 92, 92, 93, 94, 94, 95,
-	96, 96, 97, 98, 98, 99, 100, 101,
-	101, 102, 103, 104, 104, 105, 106, 107,
-	107, 108, 109, 110, 111, 111, 112, 113,
-	114, 115, 115, 116, 117, 118, 119, 120,
-	121, 121, 122, 123, 124, 125, 126, 127,
-	128, 129, 130, 131, 132, 132, 133, 134,
-	135, 136, 137, 138, 139, 140, 141, 142,
-	143, 145, 146, 147, 148, 149, 150, 151,
-	152, 153, 154, 155, 157, 158, 159, 160,
-	161, 162, 163, 165, 166, 167, 168, 169,
-	171, 172, 173, 174, 176, 177, 178, 180,
-	181, 182, 184, 185, 186, 188, 189, 190,
-	192, 193, 194, 196, 197, 199, 200, 202,
-	203, 205, 206, 208, 209, 211, 212, 214,
-	215, 217, 218, 220, 222, 223, 225, 226,
-	228, 230, 231, 233, 235, 236, 238, 240,
-	242, 243, 245, 247, 249, 251, 252, 254
-};
-
-static const byte map_gm_to_fm[128][30] = {
-	// 0x00
-{ 0xC2, 0xC5, 0x2B, 0x99, 0x58, 0xC2, 0x1F, 0x1E, 0xC8, 0x7C, 0x0A, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x23 },
-{ 0x22, 0x53, 0x0E, 0x8A, 0x30, 0x14, 0x06, 0x1D, 0x7A, 0x5C, 0x06, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0x06, 0x00, 0x1C, 0x79, 0x40, 0x02, 0x00, 0x4B, 0x79, 0x58, 0x08, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xC2, 0x89, 0x2A, 0x89, 0x49, 0xC2, 0x16, 0x1C, 0xB8, 0x7C, 0x04, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x23 },
-{ 0xC2, 0x17, 0x3D, 0x6A, 0x00, 0xC4, 0x2E, 0x2D, 0xC9, 0x20, 0x00, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0x06, 0x1E, 0x1C, 0x99, 0x00, 0x02, 0x3A, 0x4C, 0x79, 0x00, 0x0C, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0x84, 0x40, 0x3B, 0x5A, 0x6F, 0x81, 0x0E, 0x3B, 0x5A, 0x7F, 0x0B, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0x84, 0x40, 0x3B, 0x5A, 0x63, 0x81, 0x00, 0x3B, 0x5A, 0x7F, 0x01, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0x8C, 0x80, 0x05, 0xEA, 0x59, 0x82, 0x0A, 0x3C, 0xAA, 0x64, 0x07, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0x85, 0x40, 0x0D, 0xEC, 0x71, 0x84, 0x58, 0x3E, 0xCB, 0x7C, 0x01, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0x8A, 0xC0, 0x0C, 0xDC, 0x50, 0x88, 0x58, 0x3D, 0xDA, 0x7C, 0x01, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xC9, 0x40, 0x2B, 0x78, 0x42, 0xC2, 0x04, 0x4C, 0x8A, 0x7C, 0x00, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x1A },
-{ 0x2A, 0x0E, 0x17, 0x89, 0x28, 0x22, 0x0C, 0x1B, 0x09, 0x70, 0x0A, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE7, 0x9B, 0x08, 0x08, 0x26, 0xE2, 0x06, 0x0A, 0x08, 0x70, 0x0A, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xC5, 0x05, 0x00, 0xFC, 0x40, 0x84, 0x00, 0x00, 0xDC, 0x50, 0x08, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0x86, 0x40, 0x5D, 0x5A, 0x41, 0x81, 0x00, 0x0B, 0x5A, 0x7F, 0x00, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-	// 0x10
-{ 0xED, 0x00, 0x7B, 0xC8, 0x40, 0xE1, 0x99, 0x4A, 0xE9, 0x7E, 0x07, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE8, 0x4F, 0x3A, 0xD7, 0x7C, 0xE2, 0x97, 0x49, 0xF9, 0x7D, 0x05, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE1, 0x10, 0x2F, 0xF7, 0x7D, 0xF3, 0x45, 0x8F, 0xC7, 0x62, 0x07, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0x01, 0x8C, 0x9F, 0xDA, 0x70, 0xE4, 0x50, 0x9F, 0xDA, 0x6A, 0x09, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0x08, 0xD5, 0x9D, 0xA5, 0x45, 0xE2, 0x3F, 0x9F, 0xD6, 0x49, 0x07, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE5, 0x0F, 0x7D, 0xB8, 0x2E, 0xA2, 0x0F, 0x7C, 0xC7, 0x61, 0x04, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xF2, 0x2A, 0x9F, 0xDB, 0x01, 0xE1, 0x04, 0x8F, 0xD7, 0x62, 0x0A, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE4, 0x88, 0x9C, 0x50, 0x64, 0xE2, 0x18, 0x70, 0xC4, 0x7C, 0x0B, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0x02, 0xA3, 0x0D, 0xDA, 0x01, 0xC2, 0x35, 0x5D, 0x58, 0x00, 0x06, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x18 },
-{ 0x42, 0x55, 0x3E, 0xEB, 0x24, 0xD4, 0x08, 0x0D, 0xA9, 0x71, 0x04, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x18 },
-{ 0xC2, 0x00, 0x2B, 0x17, 0x51, 0xC2, 0x1E, 0x4D, 0x97, 0x7C, 0x00, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x19 },
-{ 0xC6, 0x01, 0x2D, 0xA7, 0x44, 0xC2, 0x06, 0x0E, 0xA7, 0x79, 0x06, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xC2, 0x0C, 0x06, 0x06, 0x55, 0xC2, 0x3F, 0x09, 0x86, 0x7D, 0x0A, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x0A },
-{ 0xC2, 0x2E, 0x4F, 0x77, 0x00, 0xC4, 0x08, 0x0E, 0x98, 0x59, 0x0A, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xC2, 0x30, 0x4F, 0xCA, 0x01, 0xC4, 0x0D, 0x0E, 0xB8, 0x7F, 0x08, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xC4, 0x29, 0x4F, 0xCA, 0x03, 0xC8, 0x0D, 0x0C, 0xB7, 0x7D, 0x00, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x0B },
-	// 0x20
-{ 0xC2, 0x40, 0x3C, 0x96, 0x58, 0xC4, 0xDE, 0x0E, 0xC7, 0x7C, 0x00, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x20 },
-{ 0x31, 0x13, 0x2D, 0xD7, 0x3C, 0xE2, 0x18, 0x2E, 0xB8, 0x7C, 0x08, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0x22, 0x86, 0x0D, 0xD7, 0x50, 0xE4, 0x18, 0x5E, 0xB8, 0x7C, 0x06, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x28 },
-{ 0xF2, 0x0A, 0x0D, 0xD7, 0x40, 0xE4, 0x1F, 0x5E, 0xB8, 0x7C, 0x0A, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xF2, 0x09, 0x4B, 0xD6, 0x48, 0xE4, 0x1F, 0x1C, 0xB8, 0x7C, 0x0A, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x28 },
-{ 0x62, 0x11, 0x0C, 0xE6, 0x3C, 0xE4, 0x1F, 0x0C, 0xC8, 0x7C, 0x0A, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE2, 0x12, 0x3D, 0xE6, 0x34, 0xE4, 0x1F, 0x7D, 0xB8, 0x7C, 0x0A, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE2, 0x13, 0x3D, 0xE6, 0x34, 0xE4, 0x1F, 0x5D, 0xB8, 0x7D, 0x08, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xA2, 0x40, 0x5D, 0xBA, 0x3F, 0xE2, 0x00, 0x8F, 0xD8, 0x79, 0x00, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE2, 0x40, 0x3D, 0xDA, 0x3B, 0xE1, 0x00, 0x7E, 0xD8, 0x7A, 0x04, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0x62, 0x00, 0x6D, 0xFA, 0x5D, 0xE2, 0x00, 0x8F, 0xC8, 0x79, 0x04, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE1, 0x00, 0x4E, 0xDB, 0x4A, 0xE3, 0x18, 0x6F, 0xE9, 0x7E, 0x00, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE1, 0x00, 0x4E, 0xDB, 0x66, 0xE2, 0x00, 0x7F, 0xE9, 0x7E, 0x06, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0x02, 0x0F, 0x66, 0xAA, 0x51, 0x02, 0x64, 0x29, 0xF9, 0x7C, 0x08, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x04 },
-{ 0x16, 0x4A, 0x04, 0xBA, 0x39, 0xC2, 0x58, 0x2D, 0xCA, 0x7C, 0x0A, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x03 },
-{ 0x02, 0x00, 0x01, 0x7A, 0x79, 0x02, 0x3F, 0x28, 0xEA, 0x7C, 0x08, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x02 },
-	// 0x30
-{ 0x62, 0x53, 0x9C, 0xBA, 0x31, 0x62, 0x5B, 0xAD, 0xC9, 0x55, 0x04, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xF2, 0x40, 0x6E, 0xDA, 0x49, 0xE2, 0x13, 0x8F, 0xF9, 0x7D, 0x08, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE2, 0x40, 0x8F, 0xFA, 0x50, 0xF2, 0x04, 0x7F, 0xFA, 0x7D, 0x0A, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE4, 0xA0, 0xCE, 0x5B, 0x02, 0xE2, 0x32, 0x7F, 0xFB, 0x3D, 0x04, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE6, 0x80, 0x9C, 0x99, 0x42, 0xE2, 0x04, 0x7D, 0x78, 0x60, 0x04, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xEA, 0xA0, 0xAC, 0x67, 0x02, 0xE2, 0x00, 0x7C, 0x7A, 0x7C, 0x06, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE7, 0x94, 0xAD, 0xB7, 0x03, 0xE2, 0x00, 0x7C, 0xBA, 0x7C, 0x00, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xC3, 0x3F, 0x4B, 0xE9, 0x7E, 0xC1, 0x3F, 0x9B, 0xF9, 0x7F, 0x0B, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x06 },
-{ 0xB2, 0x20, 0xAD, 0xE9, 0x00, 0x62, 0x05, 0x8F, 0xC8, 0x68, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xF2, 0x00, 0x8F, 0xFB, 0x50, 0xF6, 0x47, 0x8F, 0xE9, 0x68, 0x08, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xF2, 0x00, 0xAF, 0x88, 0x58, 0xF2, 0x54, 0x6E, 0xC9, 0x7C, 0x0A, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xF2, 0x2A, 0x9F, 0x98, 0x01, 0xE2, 0x84, 0x4E, 0x78, 0x6C, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE2, 0x02, 0x9F, 0xB8, 0x48, 0x22, 0x89, 0x9F, 0xE8, 0x7C, 0x00, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE2, 0x2A, 0x7F, 0xB8, 0x01, 0xE4, 0x00, 0x0D, 0xC5, 0x7C, 0x0C, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE4, 0x28, 0x8E, 0xE8, 0x01, 0xF2, 0x00, 0x4D, 0xD6, 0x7D, 0x0C, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0x62, 0x23, 0x8F, 0xEA, 0x00, 0xF2, 0x00, 0x5E, 0xD9, 0x7C, 0x0C, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-	// 0x40
-{ 0xB4, 0x26, 0x6E, 0x98, 0x01, 0x62, 0x00, 0x7D, 0xC8, 0x7D, 0x00, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE2, 0x2E, 0x20, 0xD9, 0x01, 0xF2, 0x0F, 0x90, 0xF8, 0x78, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE4, 0x28, 0x7E, 0xF8, 0x01, 0xE2, 0x23, 0x8E, 0xE8, 0x7D, 0x08, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xB8, 0x28, 0x9E, 0x98, 0x01, 0x62, 0x00, 0x3D, 0xC8, 0x7D, 0x08, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0x62, 0x00, 0x8E, 0xC9, 0x3D, 0xE6, 0x00, 0x7E, 0xD8, 0x68, 0x0A, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE2, 0x00, 0x5F, 0xF9, 0x48, 0xE6, 0x98, 0x8F, 0xF8, 0x7D, 0x08, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0x62, 0x0C, 0x6E, 0xD8, 0x3D, 0x2A, 0x06, 0x7D, 0xD8, 0x58, 0x04, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE4, 0x00, 0x7E, 0x89, 0x38, 0xE6, 0x84, 0x80, 0xF8, 0x68, 0x0C, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE4, 0x80, 0x6C, 0xD9, 0x30, 0xE2, 0x00, 0x8D, 0xC8, 0x7C, 0x00, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE2, 0x80, 0x88, 0x48, 0x40, 0xE2, 0x0A, 0x7D, 0xA8, 0x7C, 0x08, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE4, 0x00, 0x77, 0xC5, 0x54, 0xE2, 0x00, 0x9E, 0xD7, 0x70, 0x06, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE4, 0x80, 0x86, 0xB9, 0x64, 0xE2, 0x05, 0x9F, 0xD7, 0x78, 0x0A, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE2, 0x00, 0x68, 0x68, 0x56, 0xE2, 0x08, 0x9B, 0xB3, 0x7C, 0x08, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE4, 0x00, 0xA6, 0x87, 0x41, 0xE2, 0x0A, 0x7E, 0xC9, 0x7C, 0x06, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE4, 0x80, 0x9A, 0xB8, 0x48, 0xE2, 0x00, 0x9E, 0xF9, 0x60, 0x09, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE2, 0x80, 0x8E, 0x64, 0x68, 0xE2, 0x28, 0x6F, 0x73, 0x7C, 0x01, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-	// 0x50
-{ 0xE8, 0x00, 0x7D, 0x99, 0x54, 0xE6, 0x80, 0x80, 0xF8, 0x7C, 0x0C, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE6, 0x00, 0x9F, 0xB9, 0x6D, 0xE1, 0x00, 0x8F, 0xC8, 0x7D, 0x02, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE4, 0x00, 0x09, 0x68, 0x4A, 0xE2, 0x2B, 0x9E, 0xF3, 0x7C, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xC4, 0x00, 0x99, 0xE8, 0x3B, 0xE2, 0x25, 0x6F, 0x93, 0x7C, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE6, 0x00, 0x6F, 0xDA, 0x69, 0xE2, 0x05, 0x2F, 0xD8, 0x6A, 0x08, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xEC, 0x60, 0x9D, 0xC7, 0x00, 0xE2, 0x21, 0x7F, 0xC9, 0x7C, 0x06, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE3, 0x00, 0x0F, 0xF7, 0x7D, 0xE1, 0x3F, 0x0F, 0xA7, 0x01, 0x0D, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE4, 0xA9, 0x0F, 0xA8, 0x02, 0xE2, 0x3C, 0x5F, 0xDA, 0x3C, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE8, 0x40, 0x0D, 0x89, 0x7D, 0xE2, 0x17, 0x7E, 0xD9, 0x7C, 0x07, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE1, 0x00, 0xDF, 0x8A, 0x56, 0xE2, 0x5E, 0xCF, 0xBA, 0x7E, 0x08, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE2, 0x00, 0x0B, 0x68, 0x60, 0xE2, 0x01, 0x9E, 0xB8, 0x7C, 0x0A, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xEA, 0x00, 0xAE, 0xAB, 0x49, 0xE2, 0x00, 0xAE, 0xBA, 0x6C, 0x08, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xEB, 0x80, 0x8C, 0xCB, 0x3A, 0xE2, 0x86, 0xAF, 0xCA, 0x7C, 0x08, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE5, 0x40, 0xDB, 0x3B, 0x3C, 0xE2, 0x80, 0xBE, 0xCA, 0x71, 0x00, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE4, 0x00, 0x9E, 0xAA, 0x3D, 0xE1, 0x43, 0x0F, 0xBA, 0x7E, 0x04, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE7, 0x40, 0xEC, 0xCA, 0x44, 0xE2, 0x03, 0xBF, 0xBA, 0x66, 0x02, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-	// 0x60
-{ 0xEA, 0x00, 0x68, 0xB8, 0x48, 0xE2, 0x0A, 0x8E, 0xB8, 0x7C, 0x0C, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0x61, 0x00, 0xBE, 0x99, 0x7E, 0xE3, 0x40, 0xCF, 0xCA, 0x7D, 0x09, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xCD, 0x00, 0x0B, 0x00, 0x48, 0xC2, 0x58, 0x0C, 0x00, 0x7C, 0x0C, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x1C },
-{ 0xE2, 0x00, 0x0E, 0x00, 0x52, 0xE2, 0x58, 0x5F, 0xD0, 0x7D, 0x08, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xCC, 0x00, 0x7D, 0xDA, 0x40, 0xC2, 0x00, 0x5E, 0x9B, 0x58, 0x0C, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE9, 0xC0, 0xEE, 0xD8, 0x43, 0xE2, 0x05, 0xDD, 0xAA, 0x70, 0x06, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xDA, 0x00, 0x8F, 0xAC, 0x4A, 0x22, 0x05, 0x8D, 0x8A, 0x75, 0x02, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0x62, 0x8A, 0xCB, 0x7A, 0x74, 0xE6, 0x56, 0xAF, 0xDB, 0x70, 0x02, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xC2, 0x41, 0xAC, 0x5B, 0x5B, 0xC2, 0x80, 0x0D, 0xCB, 0x7D, 0x0A, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x12 },
-{ 0x75, 0x00, 0x0E, 0xCB, 0x5A, 0xE2, 0x1E, 0x0A, 0xC9, 0x7D, 0x0A, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x10 },
-{ 0x41, 0x00, 0x0E, 0xEA, 0x53, 0xC2, 0x00, 0x08, 0xCA, 0x7C, 0x08, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x07 },
-{ 0xC1, 0x40, 0x0C, 0x59, 0x6A, 0xC2, 0x80, 0x3C, 0xAB, 0x7C, 0x08, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x0D },
-{ 0x4B, 0x00, 0x0A, 0xF5, 0x61, 0xC2, 0x19, 0x0C, 0xE9, 0x7C, 0x08, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x07 },
-{ 0x62, 0x00, 0x7F, 0xD8, 0x54, 0xEA, 0x00, 0x8F, 0xD8, 0x7D, 0x0A, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE1, 0x00, 0x7F, 0xD9, 0x56, 0xE1, 0x00, 0x8F, 0xD8, 0x7E, 0x06, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0xE1, 0x00, 0x7F, 0xD9, 0x56, 0xE1, 0x00, 0x8F, 0xD8, 0x7E, 0x06, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-	// 0x70
-{ 0xCF, 0x40, 0x09, 0xEA, 0x54, 0xC4, 0x00, 0x0C, 0xDB, 0x64, 0x08, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x02 },
-{ 0xCF, 0x40, 0x0C, 0xAA, 0x54, 0xC4, 0x00, 0x18, 0xF9, 0x64, 0x0C, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x02 },
-{ 0xC9, 0x0E, 0x88, 0xD9, 0x3E, 0xC2, 0x08, 0x1A, 0xEA, 0x6C, 0x0C, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x05 },
-{ 0x03, 0x00, 0x15, 0x00, 0x64, 0x02, 0x00, 0x08, 0x00, 0x7C, 0x09, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x02 },
-{ 0x01, 0x00, 0x47, 0xD7, 0x6C, 0x01, 0x3F, 0x0C, 0xFB, 0x7C, 0x0A, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x04 },
-{ 0x00, 0x00, 0x36, 0x67, 0x7C, 0x01, 0x3F, 0x0E, 0xFA, 0x7C, 0x00, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x05 },
-{ 0x02, 0x00, 0x36, 0x68, 0x7C, 0x01, 0x3F, 0x0E, 0xFA, 0x7C, 0x00, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x05 },
-{ 0xCB, 0x00, 0xAF, 0x00, 0x7E, 0xC0, 0x00, 0xC0, 0x06, 0x7F, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x0F },
-{ 0x05, 0x0D, 0x80, 0xA6, 0x7F, 0x0B, 0x38, 0xA9, 0xD8, 0x00, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x04 },
-{ 0x0F, 0x00, 0x90, 0xFA, 0x68, 0x06, 0x00, 0xA7, 0x39, 0x54, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x06 },
-{ 0xC9, 0x15, 0xDD, 0xFF, 0x7C, 0x00, 0x00, 0xE7, 0xFC, 0x6C, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x38 },
-{ 0x48, 0x3C, 0x30, 0xF6, 0x03, 0x0A, 0x38, 0x97, 0xE8, 0x00, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x04 },
-{ 0x07, 0x80, 0x0B, 0xC8, 0x65, 0x02, 0x3F, 0x0C, 0xEA, 0x7C, 0x0F, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x05 },
-{ 0x00, 0x21, 0x66, 0x40, 0x03, 0x00, 0x3F, 0x47, 0x00, 0x00, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x02 },
-{ 0x08, 0x00, 0x0B, 0x3C, 0x7C, 0x08, 0x3F, 0x06, 0xF3, 0x00, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x02 },
-{ 0x00, 0x3F, 0x4C, 0xFB, 0x00, 0x00, 0x3F, 0x0A, 0xE9, 0x7C, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x05 }
-};
-
-static byte gm_percussion_to_fm[39][30] = {
-{ 0x1A, 0x3F, 0x15, 0x05, 0x7C, 0x02, 0x21, 0x2B, 0xE4, 0x7C, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x06 },
-{ 0x11, 0x12, 0x04, 0x07, 0x7C, 0x02, 0x23, 0x0B, 0xE5, 0x7C, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x05 },
-{ 0x0A, 0x3F, 0x0B, 0x01, 0x7C, 0x1F, 0x1C, 0x46, 0xD0, 0x7C, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x01 },
-{ 0x00, 0x3F, 0x0F, 0x00, 0x7C, 0x10, 0x12, 0x07, 0x00, 0x7C, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x02 },
-{ 0x0F, 0x3F, 0x0B, 0x00, 0x7C, 0x1F, 0x0F, 0x19, 0xD0, 0x7C, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x02 },
-{ 0x00, 0x3F, 0x1F, 0x00, 0x7E, 0x1F, 0x16, 0x07, 0x00, 0x7C, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x03 },
-{ 0x12, 0x3F, 0x05, 0x06, 0x7C, 0x03, 0x1F, 0x4A, 0xD9, 0x7C, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x03 },
-{ 0xCF, 0x7F, 0x08, 0xFF, 0x7E, 0x00, 0xC7, 0x2D, 0xF7, 0x73, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x02 },
-{ 0x12, 0x3F, 0x05, 0x06, 0x7C, 0x43, 0x21, 0x0C, 0xE9, 0x7C, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x03 },
-{ 0xCF, 0x7F, 0x08, 0xCF, 0x7E, 0x00, 0x45, 0x2A, 0xF8, 0x4B, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x0C },
-{ 0x12, 0x3F, 0x06, 0x17, 0x7C, 0x03, 0x27, 0x0B, 0xE9, 0x7C, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x03 },
-{ 0xCF, 0x7F, 0x08, 0xCD, 0x7E, 0x00, 0x40, 0x1A, 0x69, 0x63, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x0C },
-{ 0x13, 0x3F, 0x05, 0x06, 0x7C, 0x03, 0x17, 0x0A, 0xD9, 0x7C, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x03 },
-{ 0x15, 0x3F, 0x05, 0x06, 0x7C, 0x03, 0x21, 0x0C, 0xE9, 0x7C, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x03 },
-{ 0xCF, 0x3F, 0x2B, 0xFB, 0x7E, 0xC0, 0x1E, 0x1A, 0xCA, 0x7F, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x10 },
-{ 0x17, 0x3F, 0x04, 0x09, 0x7C, 0x03, 0x22, 0x0D, 0xE9, 0x7C, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x03 },
-{ 0xCF, 0x3F, 0x0F, 0x5E, 0x7C, 0xC6, 0x13, 0x00, 0xCA, 0x7F, 0x04, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x03 },
-{ 0xCF, 0x3F, 0x7E, 0x9D, 0x7C, 0xC8, 0xC0, 0x0A, 0xBA, 0x74, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x06 },
-{ 0xCF, 0x3F, 0x4D, 0x9F, 0x7C, 0xC6, 0x00, 0x08, 0xDA, 0x5B, 0x04, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x04 },
-{ 0xCF, 0x3F, 0x5D, 0xAA, 0x7A, 0xC0, 0xA4, 0x67, 0x99, 0x7C, 0x0A, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x02 },
-{ 0xCF, 0x3F, 0x4A, 0xFD, 0x7C, 0xCF, 0x00, 0x59, 0xEA, 0x7C, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x02 },
-{ 0x0F, 0x18, 0x0A, 0xFA, 0x57, 0x06, 0x07, 0x06, 0x39, 0x7C, 0x0A, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x02 },
-{ 0xCF, 0x3F, 0x2B, 0xFC, 0x7C, 0xCC, 0xC6, 0x0B, 0xEA, 0x7F, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x10 },
-{ 0x05, 0x1A, 0x04, 0x00, 0x7C, 0x12, 0x10, 0x0C, 0xEA, 0x7C, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x07 },
-{ 0x04, 0x19, 0x04, 0x00, 0x7C, 0x12, 0x10, 0x2C, 0xEA, 0x7C, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x04 },
-{ 0x04, 0x0A, 0x04, 0x00, 0x6C, 0x01, 0x07, 0x0D, 0xFA, 0x74, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x07 },
-{ 0x15, 0x14, 0x05, 0x00, 0x7D, 0x01, 0x07, 0x5C, 0xE9, 0x7C, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x05 },
-{ 0x10, 0x10, 0x05, 0x08, 0x7C, 0x01, 0x08, 0x0D, 0xEA, 0x7C, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x05 },
-{ 0x11, 0x00, 0x06, 0x87, 0x7F, 0x02, 0x40, 0x09, 0x59, 0x68, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x08 },
-{ 0x13, 0x26, 0x04, 0x6A, 0x7F, 0x01, 0x00, 0x08, 0x5A, 0x7C, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x08 },
-{ 0xCF, 0x4E, 0x0C, 0xAA, 0x50, 0xC4, 0x00, 0x18, 0xF9, 0x54, 0x04, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x02 },
-{ 0xCF, 0x4E, 0x0C, 0xAA, 0x50, 0xC3, 0x00, 0x18, 0xF8, 0x54, 0x04, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x02 },
-{ 0xCB, 0x3F, 0x8F, 0x00, 0x7E, 0xC5, 0x00, 0x98, 0xD6, 0x5F, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x0D },
-{ 0x0C, 0x18, 0x87, 0xB3, 0x7F, 0x19, 0x10, 0x55, 0x75, 0x7C, 0x0E, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x02 },
-{ 0x05, 0x11, 0x15, 0x00, 0x64, 0x02, 0x08, 0x08, 0x00, 0x5C, 0x09, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x02 },
-{ 0x04, 0x08, 0x15, 0x00, 0x48, 0x01, 0x08, 0x08, 0x00, 0x60, 0x08, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x02 },
-{ 0xDA, 0x00, 0x53, 0x30, 0x68, 0x07, 0x1E, 0x49, 0xC4, 0x7E, 0x03, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 },
-{ 0x1C, 0x00, 0x07, 0xBC, 0x6C, 0x0C, 0x14, 0x0B, 0x6A, 0x7E, 0x0B, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x03 },
-{ 0x0A, 0x0E, 0x7F, 0x00, 0x7D, 0x13, 0x20, 0x28, 0x03, 0x7C, 0x06, 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0x00 }
-};
-
-static const byte gm_percussion_lookup[128] = {
-	0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
-	0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
-	0xFF, 0xFF, 0xFF, 0x00, 0x01, 0x02, 0x03, 0x04, 0x05, 0x06, 0x07, 0x08, 0x09, 0x0A, 0x0B, 0x0C,
-	0x0D, 0x0E, 0x0F, 0x10, 0x11, 0x12, 0x13, 0x14, 0x15, 0x16, 0xFF, 0xFF, 0x17, 0x18, 0x19, 0x1A,
-	0x1B, 0x1C, 0x1D, 0x1E, 0x1F, 0x20, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0x21, 0x22, 0x23, 0xFF, 0xFF,
-	0x24, 0x25, 0xFF, 0xFF, 0xFF, 0x26, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
-	0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
-	0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF
-};
-
-static byte lookup_table[64][32];
-
-static const byte volume_table[] = {
-	0, 4, 7, 11,
-	13, 16, 18, 20,
-	22, 24, 26, 27,
-	29, 30, 31, 33,
-	34, 35, 36, 37,
-	38, 39, 40, 41,
-	42, 43, 44, 44,
-	45, 46, 47, 47,
-	48, 49, 49, 50,
-	51, 51, 52, 53,
-	53, 54, 54, 55,
-	55, 56, 56, 57,
-	57, 58, 58, 59,
-	59, 60, 60, 60,
-	61, 61, 62, 62,
-	62, 63, 63, 63
-};
-
-static int lookup_volume(int a, int b) {
-	if (b == 0)
-		return 0;
-
-	if (b == 31)
-		return a;
-
-	if (a < -63 || a > 63) {
-		return b * (a + 1) >> 5;
-	}
-
-	if (b < 0) {
-		if (a < 0) {
-			return lookup_table[-a][-b];
-		} else {
-			return -lookup_table[a][-b];
-		}
-	} else {
-		if (a < 0) {
-			return -lookup_table[-a][b];
-		} else {
-			return lookup_table[a][b];
-		}
-	}
-}
-
-static void create_lookup_table() {
-	int i, j;
-	int sum;
-
-	for (i = 0; i < 64; i++) {
-		sum = i;
-		for (j = 0; j < 32; j++) {
-			lookup_table[i][j] = sum >> 5;
-			sum += i;
-		}
-	}
-	for (i = 0; i < 64; i++)
-		lookup_table[i][0] = 0;
-}
-
-////////////////////////////////////////
-//
-// AdLib MIDI driver
-//
-////////////////////////////////////////
-
-class MidiDriver_ADLIB : public MidiDriver_Emulated {
-	friend class AdLibPart;
-	friend class AdLibPercussionChannel;
-
-public:
-	MidiDriver_ADLIB(Audio::Mixer *mixer);
-
-	int open();
-	void close();
-	void send(uint32 b);
-	void send(byte channel, uint32 b); // Supports higher than channel 15
-	uint32 property(int prop, uint32 param);
-
-	void setPitchBendRange(byte channel, uint range);
-	void sysEx_customInstrument(byte channel, uint32 type, const byte *instr);
-
-	MidiChannel *allocateChannel();
-	MidiChannel *getPercussionChannel() { return &_percussion; } // Percussion partially supported
-
-
-	// AudioStream API
-	bool isStereo() const { return false; }
-	int getRate() const { return _mixer->getOutputRate(); }
-
-private:
-	bool _scummSmallHeader;	// FIXME: This flag controls a special mode for SCUMM V3 games
-
-	FM_OPL *_opl;
-	byte *_adlib_reg_cache;
-
-	int _adlib_timer_counter;
-
-	uint16 channel_table_2[9];
-	int _voice_index;
-	int _timer_p;
-	int _timer_q;
-	uint16 curnote_table[9];
-	AdLibVoice _voices[9];
-	AdLibPart _parts[32];
-	AdLibPercussionChannel _percussion;
-
-	void generateSamples(int16 *buf, int len);
-	void onTimer();
-	void part_key_on(AdLibPart *part, AdLibInstrument *instr, byte note, byte velocity);
-	void part_key_off(AdLibPart *part, byte note);
-
-	void adlib_key_off(int chan);
-	void adlib_note_on(int chan, byte note, int mod);
-	void adlib_note_on_ex(int chan, byte note, int mod);
-	int adlib_get_reg_value_param(int chan, byte data);
-	void adlib_setup_channel(int chan, AdLibInstrument * instr, byte vol_1, byte vol_2);
-	byte adlib_get_reg_value(byte reg) {
-		return _adlib_reg_cache[reg];
-	}
-	void adlib_set_param(int channel, byte param, int value);
-	void adlib_key_onoff(int channel);
-	void adlib_write(byte reg, byte value);
-	void adlib_playnote(int channel, int note);
-
-	AdLibVoice *allocate_voice(byte pri);
-
-	void mc_off(AdLibVoice * voice);
-
-	static void link_mc(AdLibPart *part, AdLibVoice *voice);
-	void mc_inc_stuff(AdLibVoice *voice, Struct10 * s10, Struct11 * s11);
-	void mc_init_stuff(AdLibVoice *voice, Struct10 * s10, Struct11 * s11, byte flags,
-						InstrumentExtra * ie);
-
-	void struct10_init(Struct10 * s10, InstrumentExtra * ie);
-	static byte struct10_ontimer(Struct10 * s10, Struct11 * s11);
-	static void struct10_setup(Struct10 * s10);
-	static int random_nr(int a);
-	void mc_key_on(AdLibVoice *voice, AdLibInstrument *instr, byte note, byte velocity);
-};
-
-// MidiChannel method implementations
-
-void AdLibPart::init(MidiDriver_ADLIB *owner, byte channel) {
-	_owner = owner;
-	_channel = channel;
-	_pri_eff = 127;
-	programChange(0);
-}
-
-MidiDriver *AdLibPart::device() {
-	return _owner;
-}
-
-void AdLibPart::send(uint32 b) {
-	_owner->send(_channel, b);
-}
-
-void AdLibPart::noteOff(byte note) {
-#ifdef DEBUG_ADLIB
-	debug(6, "%10d: noteOff(%d)", tick, note);
-#endif
-	_owner->part_key_off(this, note);
-}
-
-void AdLibPart::noteOn(byte note, byte velocity) {
-#ifdef DEBUG_ADLIB
-	debug(6, "%10d: noteOn(%d,%d)", tick, note, velocity);
-#endif
-	_owner->part_key_on(this, &_part_instr, note, velocity);
-}
-
-void AdLibPart::programChange(byte program) {
-	if (program > 127)
-		return;
-
-	uint i;
-	uint count = 0;
-	for (i = 0; i < ARRAYSIZE(map_gm_to_fm[0]); ++i)
-		count += map_gm_to_fm[program][i];
-	if (!count)
-		warning("No AdLib instrument defined for GM program %d", (int) program);
-	_program = program;
-	memcpy(&_part_instr, &map_gm_to_fm[program], sizeof(AdLibInstrument));
-}
-
-void AdLibPart::pitchBend(int16 bend) {
-	AdLibVoice *voice;
-
-	_pitchbend = bend;
-	for (voice = _voice; voice; voice = voice->_next) {
-		_owner->adlib_note_on(voice->_channel, voice->_note + _transpose_eff,
-						(_pitchbend * _pitchbend_factor >> 6) + _detune_eff);
-	}
-}
-
-void AdLibPart::controlChange(byte control, byte value) {
-	switch (control) {
-	case 0:
-	case 32:
-		break; // Bank select. Not supported
-	case 1:   modulationWheel(value); break;
-	case 7:   volume(value); break;
-	case 10:  break; // Pan position. Not supported.
-	case 16:  pitchBendFactor(value); break;
-	case 17:  detune(value); break;
-	case 18:  priority(value); break;
-	case 64:  sustain(value > 0); break;
-	case 91:  break; // Effects level. Not supported.
-	case 93:  break; // Chorus level. Not supported.
-	case 119: break; // Unknown, used in Simon the Sorcerer 2
-	case 121: // reset all controllers
-		modulationWheel(0);
-		pitchBendFactor(0);
-		detune(0);
-		sustain(0);
-		break;
-	case 123: allNotesOff(); break;
-	default:
-		warning("AdLib: Unknown control change message %d (%d)", (int) control, (int)value);
-	}
-}
-
-void AdLibPart::modulationWheel(byte value) {
-	AdLibVoice *voice;
-
-	_modwheel = value;
-	for (voice = _voice; voice; voice = voice->_next) {
-		if (voice->_s10a.active && voice->_s11a.flag0x40)
-			voice->_s10a.modwheel = _modwheel >> 2;
-		if (voice->_s10b.active && voice->_s11b.flag0x40)
-			voice->_s10b.modwheel = _modwheel >> 2;
-	}
-}
-
-void AdLibPart::volume(byte value) {
-	AdLibVoice *voice;
-
-	_vol_eff = value;
-	for (voice = _voice; voice; voice = voice->_next) {
-		_owner->adlib_set_param(voice->_channel, 0, volume_table[lookup_table[voice->_vol_2][_vol_eff >> 2]]);
-		if (voice->_twochan) {
-			_owner->adlib_set_param(voice->_channel, 13, volume_table[lookup_table[voice->_vol_1][_vol_eff >> 2]]);
-		}
-	}
-}
-
-void AdLibPart::pitchBendFactor(byte value) {
-	AdLibVoice *voice;
-
-	_pitchbend_factor = value;
-	for (voice = _voice; voice; voice = voice->_next) {
-		_owner->adlib_note_on(voice->_channel, voice->_note + _transpose_eff,
-							(_pitchbend * _pitchbend_factor >> 6) + _detune_eff);
-	}
-}
-
-void AdLibPart::detune(byte value) {
-	AdLibVoice *voice;
-
-	_detune_eff = value;
-	for (voice = _voice; voice; voice = voice->_next) {
-		_owner->adlib_note_on(voice->_channel, voice->_note + _transpose_eff,
-						(_pitchbend * _pitchbend_factor >> 6) + _detune_eff);
-	}
-}
-
-void AdLibPart::priority(byte value) {
-	_pri_eff = value;
-}
-
-void AdLibPart::sustain(bool value) {
-	AdLibVoice *voice;
-
-	_pedal = value;
-	if (!value) {
-		for (voice = _voice; voice; voice = voice->_next) {
-			if (voice->_waitforpedal)
-				_owner->mc_off(voice);
-		}
-	}
-}
-
-void AdLibPart::allNotesOff() {
-	while (_voice)
-		_owner->mc_off(_voice);
-}
-
-void AdLibPart::sysEx_customInstrument(uint32 type, const byte *instr) {
-	if (type == 'ADL ') {
-		AdLibInstrument *i = &_part_instr;
-		memcpy(i, instr, sizeof(AdLibInstrument));
-	}
-}
-
-// MidiChannel method implementations for percussion
-
-AdLibPercussionChannel::~AdLibPercussionChannel() {
-	for (int i = 0; i < ARRAYSIZE(_customInstruments); ++i) {
-		delete _customInstruments[i];
-	}
-}
-
-void AdLibPercussionChannel::init(MidiDriver_ADLIB *owner, byte channel) {
-	AdLibPart::init(owner, channel);
-	_pri_eff = 0;
-	_vol_eff = 127;
-
-	// Initialize the custom instruments data
-	memset(_notes, 0, sizeof(_notes));
-	memset(_customInstruments, 0, sizeof(_customInstruments));
-}
-
-void AdLibPercussionChannel::noteOff(byte note) {
-	// Jamieson630: Unless I run into a specific instrument that
-	// may require a key off, I'm going to ignore this message.
-	// The rationale is that a percussion instrument should
-	// fade out of its own accord, and the AdLib instrument
-	// definitions used should follow this rule. Since
-	// percussion voices are allocated at the lowest priority
-	// anyway, we know that "hanging" percussion sounds will
-	// not prevent later musical instruments (or even other
-	// percussion sounds) from playing.
-/*
-	_owner->part_key_off(this, note);
-*/
-}
-
-void AdLibPercussionChannel::noteOn(byte note, byte velocity) {
-	AdLibInstrument *inst = NULL;
-
-	// The custom instruments have priority over the default mapping
-	inst = _customInstruments[note];
-	if (inst)
-		note = _notes[note];
-
-	if (!inst) {
-		// Use the default GM to FM mapping as a fallback as a fallback
-		byte key = gm_percussion_lookup[note];
-		if (key != 0xFF)
-			inst = (AdLibInstrument *)&gm_percussion_to_fm[key];
-	}
-
-	if (!inst) {
-		debug(2, "No instrument FM definition for GM percussion key %d", (int)note);
-		return;
-	}
-
-	_owner->part_key_on(this, inst, note, velocity);
-}
-
-void AdLibPercussionChannel::sysEx_customInstrument(uint32 type, const byte *instr) {
-	if (type == 'ADLP') {
-		byte note = instr[0];
-		_notes[note] = instr[1];
-
-		// Allocate memory for the new instruments
-		if (!_customInstruments[note]) {
-			_customInstruments[note] = new AdLibInstrument;
-		}
-
-		// Save the new instrument data
-		_customInstruments[note]->mod_characteristic     = instr[2];
-		_customInstruments[note]->mod_scalingOutputLevel = instr[3];
-		_customInstruments[note]->mod_attackDecay        = instr[4];
-		_customInstruments[note]->mod_sustainRelease     = instr[5];
-		_customInstruments[note]->mod_waveformSelect     = instr[6];
-		_customInstruments[note]->car_characteristic     = instr[7];
-		_customInstruments[note]->car_scalingOutputLevel = instr[8];
-		_customInstruments[note]->car_attackDecay        = instr[9];
-		_customInstruments[note]->car_sustainRelease     = instr[10];
-		_customInstruments[note]->car_waveformSelect     = instr[11];
-		_customInstruments[note]->feedback               = instr[12];
-	}
-}
-
-// MidiDriver method implementations
-
-MidiDriver_ADLIB::MidiDriver_ADLIB(Audio::Mixer *mixer)
-	: MidiDriver_Emulated(mixer) {
-	uint i;
-
-	_scummSmallHeader = false;
-
-	_adlib_reg_cache = 0;
-
-	_adlib_timer_counter = 0;
-	_voice_index = 0;
-	for (i = 0; i < ARRAYSIZE(curnote_table); ++i) {
-		curnote_table[i] = 0;
-	}
-
-	for (i = 0; i < ARRAYSIZE(_parts); ++i) {
-		_parts[i].init(this, i + ((i >= 9) ? 1 : 0));
-	}
-	_percussion.init(this, 9);
-	_timer_p = 0xD69;
-	_timer_q = 0x411B;
-}
-
-int MidiDriver_ADLIB::open() {
-	if (_isOpen)
-		return MERR_ALREADY_OPEN;
-
-	MidiDriver_Emulated::open();
-
-	int i;
-	AdLibVoice *voice;
-
-	for (i = 0, voice = _voices; i != ARRAYSIZE(_voices); i++, voice++) {
-		voice->_channel = i;
-		voice->_s11a.s10 = &voice->_s10b;
-		voice->_s11b.s10 = &voice->_s10a;
-	}
-
-	_adlib_reg_cache = (byte *)calloc(256, 1);
-
-	_opl = makeAdLibOPL(getRate());
-
-	adlib_write(1, 0x20);
-	adlib_write(8, 0x40);
-	adlib_write(0xBD, 0x00);
-	create_lookup_table();
-
-	_mixer->playStream(Audio::Mixer::kPlainSoundType, &_mixerSoundHandle, this, -1, Audio::Mixer::kMaxChannelVolume, 0, DisposeAfterUse::NO, true);
-
-	return 0;
-}
-
-void MidiDriver_ADLIB::close() {
-	if (!_isOpen)
-		return;
-	_isOpen = false;
-
-	_mixer->stopHandle(_mixerSoundHandle);
-
-	uint i;
-	for (i = 0; i < ARRAYSIZE(_voices); ++i) {
-		if (_voices[i]._part)
-			mc_off(&_voices[i]);
-	}
-
-	// Turn off the OPL emulation
-	OPLDestroy(_opl);
-//	YM3812Shutdown();
-
-	free(_adlib_reg_cache);
-}
-
-void MidiDriver_ADLIB::send(uint32 b) {
-	send(b & 0xF, b & 0xFFFFFFF0);
-}
-
-void MidiDriver_ADLIB::send(byte chan, uint32 b) {
-	//byte param3 = (byte) ((b >> 24) & 0xFF);
-	byte param2 = (byte) ((b >> 16) & 0xFF);
-	byte param1 = (byte) ((b >>  8) & 0xFF);
-	byte cmd    = (byte) (b & 0xF0);
-
-	AdLibPart *part;
-	if (chan == 9)
-		part = &_percussion;
-	else
-		part = &_parts[chan];
-
-	switch (cmd) {
-	case 0x80:// Note Off
-		part->noteOff(param1);
-		break;
-	case 0x90: // Note On
-		part->noteOn(param1, param2);
-		break;
-	case 0xA0: // Aftertouch
-		break; // Not supported.
-	case 0xB0: // Control Change
-		part->controlChange(param1, param2);
-		break;
-	case 0xC0: // Program Change
-		part->programChange(param1);
-		break;
-	case 0xD0: // Channel Pressure
-		break; // Not supported.
-	case 0xE0: // Pitch Bend
-		part->pitchBend((param1 | (param2 << 7)) - 0x2000);
-		break;
-	case 0xF0: // SysEx
-		// We should never get here! SysEx information has to be
-		// sent via high-level semantic methods.
-		warning("MidiDriver_ADLIB: Receiving SysEx command on a send() call");
-		break;
-
-	default:
-		warning("MidiDriver_ADLIB: Unknown send() command 0x%02X", cmd);
-	}
-}
-
-uint32 MidiDriver_ADLIB::property(int prop, uint32 param) {
-	switch (prop) {
-		case PROP_OLD_ADLIB: // Older games used a different operator volume algorithm
-			_scummSmallHeader = (param > 0);
-			if (_scummSmallHeader) {
-				_timer_p = 473;
-				_timer_q = 1000;
-			} else {
-				_timer_p = 0xD69;
-				_timer_q = 0x411B;
-			}
-			return 1;
-	}
-
-	return 0;
-}
-
-void MidiDriver_ADLIB::setPitchBendRange(byte channel, uint range) {
-	AdLibVoice *voice;
-	AdLibPart *part = &_parts[channel];
-
-	part->_pitchbend_factor = range;
-	for (voice = part->_voice; voice; voice = voice->_next) {
-		adlib_note_on(voice->_channel, voice->_note + part->_transpose_eff,
-					(part->_pitchbend * part->_pitchbend_factor >> 6) + part->_detune_eff);
-	}
-}
-
-void MidiDriver_ADLIB::sysEx_customInstrument(byte channel, uint32 type, const byte *instr) {
-	_parts[channel].sysEx_customInstrument(type, instr);
-}
-
-MidiChannel *MidiDriver_ADLIB::allocateChannel() {
-	AdLibPart *part;
-	uint i;
-
-	for (i = 0; i < ARRAYSIZE(_parts); ++i) {
-		part = &_parts[i];
-		if (!part->_allocated) {
-			part->allocate();
-			return part;
-		}
-	}
-	return NULL;
-}
-
-// All the code brought over from IMuseAdLib
-
-void MidiDriver_ADLIB::adlib_write(byte reg, byte value) {
-	if (_adlib_reg_cache[reg] == value)
-		return;
-#ifdef DEBUG_ADLIB
-	debug(6, "%10d: adlib_write[%x] = %x", tick, reg, value);
-#endif
-	_adlib_reg_cache[reg] = value;
-
-	OPLWriteReg(_opl, reg, value);
-}
-
-void MidiDriver_ADLIB::generateSamples(int16 *data, int len) {
-	memset(data, 0, sizeof(int16) * len);
-	YM3812UpdateOne(_opl, data, len);
-}
-
-void MidiDriver_ADLIB::onTimer() {
-	AdLibVoice *voice;
-	int i;
-
-	_adlib_timer_counter += _timer_p;
-	while (_adlib_timer_counter >= _timer_q) {
-		_adlib_timer_counter -= _timer_q;
-#ifdef DEBUG_ADLIB
-		tick++;
-#endif
-		voice = _voices;
-		for (i = 0; i != ARRAYSIZE(_voices); i++, voice++) {
-			if (!voice->_part)
-				continue;
-			if (voice->_duration && (voice->_duration -= 0x11) <= 0) {
-				mc_off(voice);
-				return;
-			}
-			if (voice->_s10a.active) {
-				mc_inc_stuff(voice, &voice->_s10a, &voice->_s11a);
-			}
-			if (voice->_s10b.active) {
-				mc_inc_stuff(voice, &voice->_s10b, &voice->_s11b);
-			}
-		}
-	}
-}
-
-void MidiDriver_ADLIB::mc_off(AdLibVoice *voice) {
-	AdLibVoice *tmp;
-
-	adlib_key_off(voice->_channel);
-
-	tmp = voice->_prev;
-
-	if (voice->_next)
-		voice->_next->_prev = tmp;
-	if (tmp)
-		tmp->_next = voice->_next;
-	else
-		voice->_part->_voice = voice->_next;
-	voice->_part = NULL;
-}
-
-void MidiDriver_ADLIB::mc_inc_stuff(AdLibVoice *voice, Struct10 *s10, Struct11 *s11) {
-	byte code;
-	AdLibPart *part = voice->_part;
-
-	code = struct10_ontimer(s10, s11);
-
-	if (code & 1) {
-		switch (s11->param) {
-		case 0:
-			voice->_vol_2 = s10->start_value + s11->modify_val;
-			if (!_scummSmallHeader) {
-				adlib_set_param(voice->_channel, 0,
-												volume_table[lookup_table[voice->_vol_2]
-																		 [part->_vol_eff >> 2]]);
-			} else {
-				adlib_set_param(voice->_channel, 0, voice->_vol_2);
-			}
-			break;
-		case 13:
-			voice->_vol_1 = s10->start_value + s11->modify_val;
-			if (voice->_twochan && !_scummSmallHeader) {
-				adlib_set_param(voice->_channel, 13,
-												volume_table[lookup_table[voice->_vol_1]
-												[part->_vol_eff >> 2]]);
-			} else {
-				adlib_set_param(voice->_channel, 13, voice->_vol_1);
-			}
-			break;
-		case 30:
-			s11->s10->modwheel = (char)s11->modify_val;
-			break;
-		case 31:
-			s11->s10->unk3 = (char)s11->modify_val;
-			break;
-		default:
-			adlib_set_param(voice->_channel, s11->param,
-							s10->start_value + s11->modify_val);
-			break;
-		}
-	}
-
-	if (code & 2 && s11->flag0x10)
-		adlib_key_onoff(voice->_channel);
-}
-
-void MidiDriver_ADLIB::adlib_key_off(int chan){
-	byte reg = chan + 0xB0;
-	adlib_write(reg, adlib_get_reg_value(reg) & ~0x20);
-}
-
-byte MidiDriver_ADLIB::struct10_ontimer(Struct10 *s10, Struct11 *s11) {
-	byte result = 0;
-	int i;
-
-	if (s10->count && (s10->count -= 17) <= 0) {
-		s10->active = 0;
-		return 0;
-	}
-
-	i = s10->cur_val + s10->speed_hi;
-	s10->speed_lo_counter += s10->speed_lo;
-	if (s10->speed_lo_counter >= s10->speed_lo_max) {
-		s10->speed_lo_counter -= s10->speed_lo_max;
-		i += s10->direction;
-	}
-	if (s10->cur_val != i || s10->modwheel != s10->modwheel_last) {
-		s10->cur_val = i;
-		s10->modwheel_last = s10->modwheel;
-		i = lookup_volume(i, s10->modwheel_last);
-		if (i != s11->modify_val) {
-			s11->modify_val = i;
-			result = 1;
-		}
-	}
-
-	if (!--s10->num_steps) {
-		s10->active++;
-		if (s10->active > 4) {
-			if (s10->loop) {
-				s10->active = 1;
-				result |= 2;
-				struct10_setup(s10);
-			} else {
-				s10->active = 0;
-			}
-		} else {
-			struct10_setup(s10);
-		}
-	}
-
-	return result;
-}
-
-void MidiDriver_ADLIB::adlib_set_param(int channel, byte param, int value) {
-	const AdLibSetParams *as;
-	byte reg;
-
-	assert(channel >= 0 && channel < 9);
-
-	if (param <= 12) {
-		reg = channel_mappings_2[channel];
-	} else if (param <= 25) {
-		param -= 13;
-		reg = channel_mappings[channel];
-	} else if (param <= 27) {
-		param -= 13;
-		reg = channel;
-	} else if (param == 28 || param == 29) {
-		if (param == 28)
-			value -= 15;
-		else
-			value -= 383;
-		value <<= 4;
-		channel_table_2[channel] = value;
-		adlib_playnote(channel, curnote_table[channel] + value);
-		return;
-	} else {
-		return;
-	}
-
-	as = &adlib_setparam_table[param];
-	if (as->d)
-		value = as->d - value;
-	reg += as->a;
-	adlib_write(reg, (adlib_get_reg_value(reg) & ~as->c) | (((byte)value) << as->b));
-}
-
-void MidiDriver_ADLIB::adlib_key_onoff(int channel) {
-	byte val;
-	byte reg = channel + 0xB0;
-	assert(channel >= 0 && channel < 9);
-
-	val = adlib_get_reg_value(reg);
-	adlib_write(reg, val & ~0x20);
-	adlib_write(reg, val | 0x20);
-}
-
-void MidiDriver_ADLIB::struct10_setup(Struct10 *s10) {
-	int b, c, d, e, f, g, h;
-	byte t;
-
-	b = s10->unk3;
-	f = s10->active - 1;
-
-	t = s10->table_a[f];
-	e = num_steps_table[lookup_table[t & 0x7F][b]];
-	if (t & 0x80) {
-		e = random_nr(e);
-	}
-	if (e == 0)
-		e++;
-
-	s10->num_steps = s10->speed_lo_max = e;
-
-	if (f != 2) {
-		c = s10->max_value;
-		g = s10->start_value;
-		t = s10->table_b[f];
-		d = lookup_volume(c, (t & 0x7F) - 31);
-		if (t & 0x80) {
-			d = random_nr(d);
-		}
-		if (d + g > c) {
-			h = c - g;
-		} else {
-			h = d;
-			if (d + g < 0)
-				h = -g;
-		}
-		h -= s10->cur_val;
-	} else {
-		h = 0;
-	}
-
-	s10->speed_hi = h / e;
-	if (h < 0) {
-		h = -h;
-		s10->direction = -1;
-	} else {
-		s10->direction = 1;
-	}
-
-	s10->speed_lo = h % e;
-	s10->speed_lo_counter = 0;
-}
-
-void MidiDriver_ADLIB::adlib_playnote(int channel, int note) {
-	byte old, oct, notex;
-	int note2;
-	int i;
-
-	note2 = (note >> 7) - 4;
-	note2 = (note2 < 128) ? note2 : 0;
-
-	oct = (note2 / 12);
-	if (oct > 7)
-		oct = 7 << 2;
-	else
-		oct <<= 2;
-	notex = note2 % 12 + 3;
-
-	old = adlib_get_reg_value(channel + 0xB0);
-	if (old & 0x20) {
-		old &= ~0x20;
-		if (oct > old) {
-			if (notex < 6) {
-				notex += 12;
-				oct -= 4;
-			}
-		} else if (oct < old) {
-			if (notex > 11) {
-				notex -= 12;
-				oct += 4;
-			}
-		}
-	}
-
-	i = (notex << 3) + ((note >> 4) & 0x7);
-	adlib_write(channel + 0xA0, note_to_f_num[i]);
-	adlib_write(channel + 0xB0, oct | 0x20);
-}
-
-int MidiDriver_ADLIB::random_nr(int a) {
-	static byte _rand_seed = 1;
-	if (_rand_seed & 1) {
-		_rand_seed >>= 1;
-		_rand_seed ^= 0xB8;
-	} else {
-		_rand_seed >>= 1;
-	}
-	return _rand_seed * a >> 8;
-}
-
-void MidiDriver_ADLIB::part_key_off(AdLibPart *part, byte note) {
-	AdLibVoice *voice;
-
-	for (voice = part->_voice; voice; voice = voice->_next) {
-		if (voice->_note == note) {
-			if (part->_pedal)
-				voice->_waitforpedal = true;
-			else
-				mc_off(voice);
-		}
-	}
-}
-
-void MidiDriver_ADLIB::part_key_on(AdLibPart *part, AdLibInstrument *instr, byte note, byte velocity) {
-	AdLibVoice *voice;
-
-	voice = allocate_voice(part->_pri_eff);
-	if (!voice)
-		return;
-
-	link_mc(part, voice);
-	mc_key_on(voice, instr, note, velocity);
-}
-
-AdLibVoice *MidiDriver_ADLIB::allocate_voice(byte pri) {
-	AdLibVoice *ac, *best = NULL;
-	int i;
-
-	for (i = 0; i < 9; i++) {
-		if (++_voice_index >= 9)
-			_voice_index = 0;
-		ac = &_voices[_voice_index];
-		if (!ac->_part)
-			return ac;
-		if (!ac->_next) {
-			if (ac->_part->_pri_eff <= pri) {
-				pri = ac->_part->_pri_eff;
-				best = ac;
-			}
-		}
-	}
-
-	/* SCUMM V3 games don't have note priorities, first comes wins. */
-	if (_scummSmallHeader)
-		return NULL;
-
-	if (best)
-		mc_off(best);
-	return best;
-}
-
-void MidiDriver_ADLIB::link_mc(AdLibPart *part, AdLibVoice *voice) {
-	voice->_part = part;
-	voice->_next = (AdLibVoice *)part->_voice;
-	part->_voice = voice;
-	voice->_prev = NULL;
-
-	if (voice->_next)
-		voice->_next->_prev = voice;
-}
-
-void MidiDriver_ADLIB::mc_key_on(AdLibVoice *voice, AdLibInstrument *instr, byte note, byte velocity) {
-	AdLibPart *part = voice->_part;
-	int c;
-	byte vol_1, vol_2;
-
-	voice->_twochan = instr->feedback & 1;
-	voice->_note = note;
-	voice->_waitforpedal = false;
-	voice->_duration = instr->duration;
-	if (voice->_duration != 0)
-		voice->_duration *= 63;
-
-	if (!_scummSmallHeader)
-		vol_1 = (instr->mod_scalingOutputLevel & 0x3F) + lookup_table[velocity >> 1][instr->mod_waveformSelect >> 2];
-	else
-		vol_1 = 0x3f - (instr->mod_scalingOutputLevel & 0x3F);
-	if (vol_1 > 0x3F)
-		vol_1 = 0x3F;
-	voice->_vol_1 = vol_1;
-
-	if (!_scummSmallHeader)
-		vol_2 = (instr->car_scalingOutputLevel & 0x3F) + lookup_table[velocity >> 1][instr->car_waveformSelect >> 2];
-	else
-		vol_2 = 0x3f - (instr->car_scalingOutputLevel & 0x3F);
-	if (vol_2 > 0x3F)
-		vol_2 = 0x3F;
-	voice->_vol_2 = vol_2;
-
-	c = part->_vol_eff >> 2;
-
-	if (!_scummSmallHeader) {
-		vol_2 = volume_table[lookup_table[vol_2][c]];
-		if (voice->_twochan)
-			vol_1 = volume_table[lookup_table[vol_1][c]];
-	}
-
-	adlib_setup_channel(voice->_channel, instr, vol_1, vol_2);
-	adlib_note_on_ex(voice->_channel, part->_transpose_eff + note, part->_detune_eff + (part->_pitchbend * part->_pitchbend_factor >> 6));
-
-	if (instr->flags_a & 0x80) {
-		mc_init_stuff(voice, &voice->_s10a, &voice->_s11a, instr->flags_a, &instr->extra_a);
-	} else {
-		voice->_s10a.active = 0;
-	}
-
-	if (instr->flags_b & 0x80) {
-		mc_init_stuff(voice, &voice->_s10b, &voice->_s11b, instr->flags_b, &instr->extra_b);
-	} else {
-		voice->_s10b.active = 0;
-	}
-}
-
-void MidiDriver_ADLIB::adlib_setup_channel(int chan, AdLibInstrument *instr, byte vol_1, byte vol_2) {
-	byte channel;
-
-	assert(chan >= 0 && chan < 9);
-
-	channel = channel_mappings[chan];
-	adlib_write(channel + 0x20, instr->mod_characteristic);
-	adlib_write(channel + 0x40, (instr->mod_scalingOutputLevel | 0x3F) - vol_1 );
-	adlib_write(channel + 0x60, 0xff & (~instr->mod_attackDecay));
-	adlib_write(channel + 0x80, 0xff & (~instr->mod_sustainRelease));
-	adlib_write(channel + 0xE0, instr->mod_waveformSelect);
-
-	channel = channel_mappings_2[chan];
-	adlib_write(channel + 0x20, instr->car_characteristic);
-	adlib_write(channel + 0x40, (instr->car_scalingOutputLevel | 0x3F) - vol_2 );
-	adlib_write(channel + 0x60, 0xff & (~instr->car_attackDecay));
-	adlib_write(channel + 0x80, 0xff & (~instr->car_sustainRelease));
-	adlib_write(channel + 0xE0, instr->car_waveformSelect);
-
-	adlib_write((byte)chan + 0xC0, instr->feedback);
-}
-
-void MidiDriver_ADLIB::adlib_note_on_ex(int chan, byte note, int mod)
-{
-	int code;
-	assert(chan >= 0 && chan < 9);
-	code = (note << 7) + mod;
-	curnote_table[chan] = code;
-	channel_table_2[chan] = 0;
-	adlib_playnote(chan, code);
-}
-
-void MidiDriver_ADLIB::mc_init_stuff(AdLibVoice *voice, Struct10 * s10,
-									Struct11 * s11, byte flags, InstrumentExtra * ie) {
-	AdLibPart *part = voice->_part;
-	s11->modify_val = 0;
-	s11->flag0x40 = flags & 0x40;
-	s10->loop = flags & 0x20;
-	s11->flag0x10 = flags & 0x10;
-	s11->param = param_table_1[flags & 0xF];
-	s10->max_value = maxval_table[flags & 0xF];
-	s10->unk3 = 31;
-	if (s11->flag0x40) {
-		s10->modwheel = part->_modwheel >> 2;
-	} else {
-		s10->modwheel = 31;
-	}
-
-	switch (s11->param) {
-	case 0:
-		s10->start_value = voice->_vol_2;
-		break;
-	case 13:
-		s10->start_value = voice->_vol_1;
-		break;
-	case 30:
-		s10->start_value = 31;
-		s11->s10->modwheel = 0;
-		break;
-	case 31:
-		s10->start_value = 0;
-		s11->s10->unk3 = 0;
-		break;
-	default:
-		s10->start_value = adlib_get_reg_value_param(voice->_channel, s11->param);
-	}
-
-	struct10_init(s10, ie);
-}
-
-void MidiDriver_ADLIB::struct10_init(Struct10 *s10, InstrumentExtra *ie) {
-	s10->active = 1;
-	if (!_scummSmallHeader) {
-		s10->cur_val = 0;
-	} else {
-		s10->cur_val = s10->start_value;
-		s10->start_value = 0;
-	}
-	s10->modwheel_last = 31;
-	s10->count = ie->a;
-	if (s10->count)
-		s10->count *= 63;
-	s10->table_a[0] = ie->b;
-	s10->table_a[1] = ie->d;
-	s10->table_a[2] = ie->f;
-	s10->table_a[3] = ie->g;
-
-	s10->table_b[0] = ie->c;
-	s10->table_b[1] = ie->e;
-	s10->table_b[2] = 0;
-	s10->table_b[3] = ie->h;
-
-	struct10_setup(s10);
-}
-
-int MidiDriver_ADLIB::adlib_get_reg_value_param(int chan, byte param) {
-	const AdLibSetParams *as;
-	byte val;
-	byte channel;
-
-	assert(chan >= 0 && chan < 9);
-
-	if (param <= 12) {
-		channel = channel_mappings_2[chan];
-	} else if (param <= 25) {
-		param -= 13;
-		channel = channel_mappings[chan];
-	} else if (param <= 27) {
-		param -= 13;
-		channel = chan;
-	} else if (param == 28) {
-		return 0xF;
-	} else if (param == 29) {
-		return 0x17F;
-	} else {
-		return 0;
-	}
-
-	as = &adlib_setparam_table[param];
-	val = adlib_get_reg_value(channel + as->a);
-	val &= as->c;
-	val >>= as->b;
-	if (as->d)
-		val = as->d - val;
-
-	return val;
-}
-
-void MidiDriver_ADLIB::adlib_note_on(int chan, byte note, int mod) {
-	int code;
-	assert(chan >= 0 && chan < 9);
-	code = (note << 7) + mod;
-	curnote_table[chan] = code;
-	adlib_playnote(chan, (int16) channel_table_2[chan] + code);
-}
-
-
-// Plugin interface
-
-class AdLibEmuMusicPlugin : public MusicPluginObject {
-public:
-	const char *getName() const {
-		return _s("AdLib Emulator");
-	}
-
-	const char *getId() const {
-		return "adlib";
-	}
-
-	MusicDevices getDevices() const;
-	Common::Error createInstance(MidiDriver **mididriver, MidiDriver::DeviceHandle = 0) const;
-};
-
-MusicDevices AdLibEmuMusicPlugin::getDevices() const {
-	MusicDevices devices;
-	devices.push_back(MusicDevice(this, "", MT_ADLIB));
-	return devices;
-}
-
-Common::Error AdLibEmuMusicPlugin::createInstance(MidiDriver **mididriver, MidiDriver::DeviceHandle) const {
-	*mididriver = new MidiDriver_ADLIB(g_system->getMixer());
-
-	return Common::kNoError;
-}
-
-//#if PLUGIN_ENABLED_DYNAMIC(ADLIB)
-	//REGISTER_PLUGIN_DYNAMIC(ADLIB, PLUGIN_TYPE_MUSIC, AdLibEmuMusicPlugin);
-//#else
-	REGISTER_PLUGIN_STATIC(ADLIB, PLUGIN_TYPE_MUSIC, AdLibEmuMusicPlugin);
-//#endif
diff --git a/sound/softsynth/appleiigs.cpp b/sound/softsynth/appleiigs.cpp
deleted file mode 100644
index 9004d1f0ab..0000000000
--- a/sound/softsynth/appleiigs.cpp
+++ /dev/null
@@ -1,57 +0,0 @@
-/* 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 "sound/null.h"
-
-//	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 AppleIIGSMusicPlugin : public NullMusicPlugin {
-public:
-	const char *getName() const {
-		return _s("Apple II GS Emulator (NOT IMPLEMENTED)");
-	}
-
-	const char *getId() const {
-		return "appleIIgs";
-	}
-
-	MusicDevices getDevices() const;
-};
-
-MusicDevices AppleIIGSMusicPlugin::getDevices() const {
-	MusicDevices devices;
-	devices.push_back(MusicDevice(this, "", MT_APPLEIIGS));
-	return devices;
-}
-
-//#if PLUGIN_ENABLED_DYNAMIC(APPLEIIGS)
-	//REGISTER_PLUGIN_DYNAMIC(APPLEIIGS, PLUGIN_TYPE_MUSIC, AppleIIGSMusicPlugin);
-//#else
-	REGISTER_PLUGIN_STATIC(APPLEIIGS, PLUGIN_TYPE_MUSIC, AppleIIGSMusicPlugin);
-//#endif
-
diff --git a/sound/softsynth/cms.cpp b/sound/softsynth/cms.cpp
deleted file mode 100644
index b307146f14..0000000000
--- a/sound/softsynth/cms.cpp
+++ /dev/null
@@ -1,376 +0,0 @@
-/* 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 "sound/softsynth/cms.h"
-#include "sound/null.h"
-
-#include "common/textconsole.h"
-#include "common/translation.h"
-#include "common/debug.h"
-
-// CMS/Gameblaster Emulation taken from DosBox
-
-#define LEFT	0x00
-#define RIGHT	0x01
-
-static const byte envelope[8][64] = {
-	/* zero amplitude */
-	{ 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
-	  0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
-	  0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
-	  0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
-	/* maximum amplitude */
-	{15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,
-	 15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,
-	 15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,
-	 15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15, },
-	/* single decay */
-	{15,14,13,12,11,10, 9, 8, 7, 6, 5, 4, 3, 2, 1, 0,
-	  0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
-	  0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
-	  0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
-	/* repetitive decay */
-	{15,14,13,12,11,10, 9, 8, 7, 6, 5, 4, 3, 2, 1, 0,
-	 15,14,13,12,11,10, 9, 8, 7, 6, 5, 4, 3, 2, 1, 0,
-	 15,14,13,12,11,10, 9, 8, 7, 6, 5, 4, 3, 2, 1, 0,
-	 15,14,13,12,11,10, 9, 8, 7, 6, 5, 4, 3, 2, 1, 0 },
-	/* single triangular */
-	{ 0, 1, 2, 3, 4, 5, 6, 7, 8, 9,10,11,12,13,14,15,
-	 15,14,13,12,11,10, 9, 8, 7, 6, 5, 4, 3, 2, 1, 0,
-	  0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
-	0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
-	/* repetitive triangular */
-	{ 0, 1, 2, 3, 4, 5, 6, 7, 8, 9,10,11,12,13,14,15,
-	 15,14,13,12,11,10, 9, 8, 7, 6, 5, 4, 3, 2, 1, 0,
-	  0, 1, 2, 3, 4, 5, 6, 7, 8, 9,10,11,12,13,14,15,
-	 15,14,13,12,11,10, 9, 8, 7, 6, 5, 4, 3, 2, 1, 0 },
-	/* single attack */
-	{ 0, 1, 2, 3, 4, 5, 6, 7, 8, 9,10,11,12,13,14,15,
-	  0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
-	  0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
-	  0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
-	/* repetitive attack */
-	{ 0, 1, 2, 3, 4, 5, 6, 7, 8, 9,10,11,12,13,14,15,
-	  0, 1, 2, 3, 4, 5, 6, 7, 8, 9,10,11,12,13,14,15,
-	  0, 1, 2, 3, 4, 5, 6, 7, 8, 9,10,11,12,13,14,15,
-	  0, 1, 2, 3, 4, 5, 6, 7, 8, 9,10,11,12,13,14,15 }
-};
-
-static const int amplitude_lookup[16] = {
-	 0*32767/16,  1*32767/16,  2*32767/16,	3*32767/16,
-	 4*32767/16,  5*32767/16,  6*32767/16,	7*32767/16,
-	 8*32767/16,  9*32767/16, 10*32767/16, 11*32767/16,
-	12*32767/16, 13*32767/16, 14*32767/16, 15*32767/16
-};
-
-void CMSEmulator::portWrite(int port, int val) {
-	switch (port) {
-	case 0x220:
-		portWriteIntern(0, 1, val);
-		break;
-
-	case 0x221:
-		_saa1099[0].selected_reg = val & 0x1f;
-		if (_saa1099[0].selected_reg == 0x18 || _saa1099[0].selected_reg == 0x19) {
-			/* clock the envelope channels */
-			if (_saa1099[0].env_clock[0])
-				envelope(0, 0);
-			if (_saa1099[0].env_clock[1])
-				envelope(0, 1);
-		}
-		break;
-
-	case 0x222:
-		portWriteIntern(1, 1, val);
-		break;
-
-	case 0x223:
-		_saa1099[1].selected_reg = val & 0x1f;
-		if (_saa1099[1].selected_reg == 0x18 || _saa1099[1].selected_reg == 0x19) {
-			/* clock the envelope channels */
-			if (_saa1099[1].env_clock[0])
-				envelope(1, 0);
-			if (_saa1099[1].env_clock[1])
-				envelope(1, 1);
-		}
-		break;
-
-	default:
-		warning("CMSEmulator got port: 0x%X", port);
-		break;
-	}
-}
-
-void CMSEmulator::readBuffer(int16 *buffer, const int numSamples) {
-	update(0, &buffer[0], numSamples);
-	update(1, &buffer[0], numSamples);
-}
-
-void CMSEmulator::envelope(int chip, int ch) {
-	SAA1099 *saa = &_saa1099[chip];
-	if (saa->env_enable[ch]) {
-		int step, mode, mask;
-		mode = saa->env_mode[ch];
-		/* step from 0..63 and then loop in steps 32..63 */
-		step = saa->env_step[ch] = ((saa->env_step[ch] + 1) & 0x3f) | (saa->env_step[ch] & 0x20);
-
-		mask = 15;
-		if (saa->env_bits[ch])
-			mask &= ~1; 	/* 3 bit resolution, mask LSB */
-
-		saa->channels[ch*3+0].envelope[ LEFT] =
-		saa->channels[ch*3+1].envelope[ LEFT] =
-		saa->channels[ch*3+2].envelope[ LEFT] = ::envelope[mode][step] & mask;
-		if (saa->env_reverse_right[ch] & 0x01) {
-			saa->channels[ch*3+0].envelope[RIGHT] =
-			saa->channels[ch*3+1].envelope[RIGHT] =
-			saa->channels[ch*3+2].envelope[RIGHT] = (15 - ::envelope[mode][step]) & mask;
-		} else {
-			saa->channels[ch*3+0].envelope[RIGHT] =
-			saa->channels[ch*3+1].envelope[RIGHT] =
-			saa->channels[ch*3+2].envelope[RIGHT] = ::envelope[mode][step] & mask;
-		}
-	} else {
-		/* envelope mode off, set all envelope factors to 16 */
-		saa->channels[ch*3+0].envelope[ LEFT] =
-		saa->channels[ch*3+1].envelope[ LEFT] =
-		saa->channels[ch*3+2].envelope[ LEFT] =
-		saa->channels[ch*3+0].envelope[RIGHT] =
-		saa->channels[ch*3+1].envelope[RIGHT] =
-		saa->channels[ch*3+2].envelope[RIGHT] = 16;
-	}
-}
-
-void CMSEmulator::update(int chip, int16 *buffer, int length) {
-	struct SAA1099 *saa = &_saa1099[chip];
-	int j, ch;
-
-	/* if the channels are disabled we're done */
-	if (!saa->all_ch_enable) {
-		/* init output data */
-		if (chip == 0) {
-			memset(buffer, 0, sizeof(int16)*length*2);
-		}
-		return;
-	}
-
-	if (chip == 0) {
-		memset(buffer, 0, sizeof(int16)*length*2);
-	}
-
-	for (ch = 0; ch < 2; ch++) {
-		switch (saa->noise_params[ch]) {
-		case 0: saa->noise[ch].freq = 31250.0 * 2; break;
-		case 1: saa->noise[ch].freq = 15625.0 * 2; break;
-		case 2: saa->noise[ch].freq =  7812.5 * 2; break;
-		case 3: saa->noise[ch].freq = saa->channels[ch * 3].freq; break;
-		}
-	}
-
-	/* fill all data needed */
-	for (j = 0; j < length; ++j) {
-		int output_l = 0, output_r = 0;
-
-		/* for each channel */
-		for (ch = 0; ch < 6; ch++) {
-			if (saa->channels[ch].freq == 0.0)
-				saa->channels[ch].freq = (double)((2 * 15625) << saa->channels[ch].octave) /
-				(511.0 - (double)saa->channels[ch].frequency);
-
-			/* check the actual position in the square wave */
-			saa->channels[ch].counter -= saa->channels[ch].freq;
-			while (saa->channels[ch].counter < 0) {
-				/* calculate new frequency now after the half wave is updated */
-				saa->channels[ch].freq = (double)((2 * 15625) << saa->channels[ch].octave) /
-					(511.0 - (double)saa->channels[ch].frequency);
-
-				saa->channels[ch].counter += _sampleRate;
-				saa->channels[ch].level ^= 1;
-
-				/* eventually clock the envelope counters */
-				if (ch == 1 && saa->env_clock[0] == 0)
-					envelope(chip, 0);
-				if (ch == 4 && saa->env_clock[1] == 0)
-					envelope(chip, 1);
-			}
-
-			/* if the noise is enabled */
-			if (saa->channels[ch].noise_enable) {
-				/* if the noise level is high (noise 0: chan 0-2, noise 1: chan 3-5) */
-				if (saa->noise[ch/3].level & 1) {
-					/* subtract to avoid overflows, also use only half amplitude */
-					output_l -= saa->channels[ch].amplitude[ LEFT] * saa->channels[ch].envelope[ LEFT] / 16 / 2;
-					output_r -= saa->channels[ch].amplitude[RIGHT] * saa->channels[ch].envelope[RIGHT] / 16 / 2;
-				}
-			}
-
-			/* if the square wave is enabled */
-			if (saa->channels[ch].freq_enable) {
-				/* if the channel level is high */
-				if (saa->channels[ch].level & 1) {
-					output_l += saa->channels[ch].amplitude[ LEFT] * saa->channels[ch].envelope[ LEFT] / 16;
-					output_r += saa->channels[ch].amplitude[RIGHT] * saa->channels[ch].envelope[RIGHT] / 16;
-				}
-			}
-		}
-
-		for (ch = 0; ch < 2; ch++) {
-			/* check the actual position in noise generator */
-			saa->noise[ch].counter -= saa->noise[ch].freq;
-			while (saa->noise[ch].counter < 0) {
-				saa->noise[ch].counter += _sampleRate;
-				if (((saa->noise[ch].level & 0x4000) == 0) == ((saa->noise[ch].level & 0x0040) == 0) )
-					saa->noise[ch].level = (saa->noise[ch].level << 1) | 1;
-				else
-					saa->noise[ch].level <<= 1;
-			}
-		}
-		/* write sound data to the buffer */
-		buffer[j*2] += output_l / 6;
-		buffer[j*2+1] += output_r / 6;
-	}
-}
-
-void CMSEmulator::portWriteIntern(int chip, int offset, int data) {
-	SAA1099 *saa = &_saa1099[chip];
-	int reg = saa->selected_reg;
-	int ch;
-
-	switch (reg) {
-	/* channel i amplitude */
-	case 0x00:
-	case 0x01:
-	case 0x02:
-	case 0x03:
-	case 0x04:
-	case 0x05:
-		ch = reg & 7;
-		saa->channels[ch].amplitude[LEFT] = amplitude_lookup[data & 0x0f];
-		saa->channels[ch].amplitude[RIGHT] = amplitude_lookup[(data >> 4) & 0x0f];
-		break;
-
-	/* channel i frequency */
-	case 0x08:
-	case 0x09:
-	case 0x0a:
-	case 0x0b:
-	case 0x0c:
-	case 0x0d:
-		ch = reg & 7;
-		saa->channels[ch].frequency = data & 0xff;
-		break;
-
-	/* channel i octave */
-	case 0x10:
-	case 0x11:
-	case 0x12:
-		ch = (reg - 0x10) << 1;
-		saa->channels[ch + 0].octave = data & 0x07;
-		saa->channels[ch + 1].octave = (data >> 4) & 0x07;
-		break;
-
-	/* channel i frequency enable */
-	case 0x14:
-		saa->channels[0].freq_enable = data & 0x01;
-		saa->channels[1].freq_enable = data & 0x02;
-		saa->channels[2].freq_enable = data & 0x04;
-		saa->channels[3].freq_enable = data & 0x08;
-		saa->channels[4].freq_enable = data & 0x10;
-		saa->channels[5].freq_enable = data & 0x20;
-		break;
-
-	/* channel i noise enable */
-	case 0x15:
-		saa->channels[0].noise_enable = data & 0x01;
-		saa->channels[1].noise_enable = data & 0x02;
-		saa->channels[2].noise_enable = data & 0x04;
-		saa->channels[3].noise_enable = data & 0x08;
-		saa->channels[4].noise_enable = data & 0x10;
-		saa->channels[5].noise_enable = data & 0x20;
-		break;
-
-	/* noise generators parameters */
-	case 0x16:
-		saa->noise_params[0] = data & 0x03;
-		saa->noise_params[1] = (data >> 4) & 0x03;
-		break;
-
-	/* envelope generators parameters */
-	case 0x18:
-	case 0x19:
-		ch = reg - 0x18;
-		saa->env_reverse_right[ch] = data & 0x01;
-		saa->env_mode[ch] = (data >> 1) & 0x07;
-		saa->env_bits[ch] = data & 0x10;
-		saa->env_clock[ch] = data & 0x20;
-		saa->env_enable[ch] = data & 0x80;
-		/* reset the envelope */
-		saa->env_step[ch] = 0;
-		break;
-
-	/* channels enable & reset generators */
-	case 0x1c:
-		saa->all_ch_enable = data & 0x01;
-		saa->sync_state = data & 0x02;
-		if (data & 0x02) {
-			int i;
-			/* Synch & Reset generators */
-			for (i = 0; i < 6; i++) {
-				saa->channels[i].level = 0;
-				saa->channels[i].counter = 0.0;
-			}
-		}
-		break;
-
-	default:
-		// The CMS allows all registers to be written, so we just output some debug
-		// message here
-		debug(5, "CMS Unknown write to reg %x with %x",reg, data);
-	}
-}
-
-class CMSMusicPlugin : public NullMusicPlugin {
-public:
-	const char *getName() const {
-		return _s("Creative Music System Emulator");
-	}
-
-	const char *getId() const {
-		return "cms";
-	}
-
-	MusicDevices getDevices() const;
-};
-
-MusicDevices CMSMusicPlugin::getDevices() const {
-	MusicDevices devices;
-	devices.push_back(MusicDevice(this, "", MT_CMS));
-	return devices;
-}
-
-//#if PLUGIN_ENABLED_DYNAMIC(CMS)
-	//REGISTER_PLUGIN_DYNAMIC(CMS, PLUGIN_TYPE_MUSIC, CMSMusicPlugin);
-//#else
-	REGISTER_PLUGIN_STATIC(CMS, PLUGIN_TYPE_MUSIC, CMSMusicPlugin);
-//#endif
diff --git a/sound/softsynth/cms.h b/sound/softsynth/cms.h
deleted file mode 100644
index d5bb7f0a42..0000000000
--- a/sound/softsynth/cms.h
+++ /dev/null
@@ -1,92 +0,0 @@
-/* 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_SOFTSYNTH_CMS_H
-#define SOUND_SOFTSYNTH_CMS_H
-
-#include "common/scummsys.h"
-
-/* this structure defines a channel */
-struct saa1099_channel {
-	int frequency;				/* frequency (0x00..0xff) */
-	int freq_enable;			/* frequency enable */
-	int noise_enable;			/* noise enable */
-	int octave;				/* octave (0x00..0x07) */
-	int amplitude[2];			/* amplitude (0x00..0x0f) */
-	int envelope[2];			/* envelope (0x00..0x0f or 0x10 == off) */
-
-	/* vars to simulate the square wave */
-	double counter;
-	double freq;
-	int level;
-};
-
-/* this structure defines a noise channel */
-struct saa1099_noise {
-	/* vars to simulate the noise generator output */
-	double counter;
-	double freq;
-	int level;				/* noise polynomal shifter */
-};
-
-/* this structure defines a SAA1099 chip */
-struct SAA1099 {
-	int stream;				/* our stream */
-	int noise_params[2];			/* noise generators parameters */
-	int env_enable[2];			/* envelope generators enable */
-	int env_reverse_right[2];		/* envelope reversed for right channel */
-	int env_mode[2];			/* envelope generators mode */
-	int env_bits[2];			/* non zero = 3 bits resolution */
-	int env_clock[2];			/* envelope clock mode (non-zero external) */
-	int env_step[2];			/* current envelope step */
-	int all_ch_enable;			/* all channels enable */
-	int sync_state;				/* sync all channels */
-	int selected_reg;			/* selected register */
-	struct saa1099_channel channels[6];	/* channels */
-	struct saa1099_noise noise[2];		/* noise generators */
-};
-
-class CMSEmulator {
-public:
-	CMSEmulator(uint32 sampleRate) {
-		_sampleRate = sampleRate;
-		memset(_saa1099, 0, sizeof(SAA1099)*2);
-	}
-
-	~CMSEmulator() { }
-
-	void portWrite(int port, int val);
-	void readBuffer(int16 *buffer, const int numSamples);
-private:
-	uint32 _sampleRate;
-
-	SAA1099 _saa1099[2];
-
-	void envelope(int chip, int ch);
-	void update(int chip, int16 *buffer, int length);
-	void portWriteIntern(int chip, int offset, int data);
-};
-
-
-#endif
diff --git a/sound/softsynth/emumidi.h b/sound/softsynth/emumidi.h
deleted file mode 100644
index 3fb2733b71..0000000000
--- a/sound/softsynth/emumidi.h
+++ /dev/null
@@ -1,116 +0,0 @@
-/* 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_SOFTSYNTH_EMUMIDI_H
-#define SOUND_SOFTSYNTH_EMUMIDI_H
-
-#include "sound/audiostream.h"
-#include "sound/mididrv.h"
-#include "sound/mixer.h"
-
-#define FIXP_SHIFT 16
-
-class MidiDriver_Emulated : public Audio::AudioStream, public MidiDriver {
-protected:
-	bool _isOpen;
-	Audio::Mixer *_mixer;
-	Audio::SoundHandle _mixerSoundHandle;
-
-private:
-	Common::TimerManager::TimerProc _timerProc;
-	void *_timerParam;
-
-	int _nextTick;
-	int _samplesPerTick;
-
-protected:
-	virtual void generateSamples(int16 *buf, int len) = 0;
-	virtual void onTimer() {}
-
-	int _baseFreq;
-
-public:
-	MidiDriver_Emulated(Audio::Mixer *mixer) : _mixer(mixer) {
-		_isOpen = false;
-
-		_timerProc = 0;
-		_timerParam = 0;
-
-		_nextTick = 0;
-		_samplesPerTick = 0;
-
-		_baseFreq = 250;
-	}
-
-	int open() {
-		_isOpen = true;
-
-		int d = getRate() / _baseFreq;
-		int r = getRate() % _baseFreq;
-
-		// This is equivalent to (getRate() << FIXP_SHIFT) / BASE_FREQ
-		// but less prone to arithmetic overflow.
-
-		_samplesPerTick = (d << FIXP_SHIFT) + (r << FIXP_SHIFT) / _baseFreq;
-		return 0;
-	}
-
-	void setTimerCallback(void *timer_param, Common::TimerManager::TimerProc timer_proc) {
-		_timerProc = timer_proc;
-		_timerParam = timer_param;
-	}
-
-	uint32 getBaseTempo() { return 1000000 / _baseFreq; }
-
-
-	// AudioStream API
-	int readBuffer(int16 *data, const int numSamples) {
-		const int stereoFactor = isStereo() ? 2 : 1;
-		int len = numSamples / stereoFactor;
-		int step;
-
-		do {
-			step = len;
-			if (step > (_nextTick >> FIXP_SHIFT))
-				step = (_nextTick >> FIXP_SHIFT);
-
-			generateSamples(data, step);
-
-			_nextTick -= step << FIXP_SHIFT;
-			if (!(_nextTick >> FIXP_SHIFT)) {
-				if (_timerProc)
-					(*_timerProc)(_timerParam);
-				onTimer();
-				_nextTick += _samplesPerTick;
-			}
-			data += step * stereoFactor;
-			len -= step;
-		} while (len);
-
-		return numSamples;
-	}
-	bool endOfData() const { return false; }
-};
-
-#endif
diff --git a/sound/softsynth/fluidsynth.cpp b/sound/softsynth/fluidsynth.cpp
deleted file mode 100644
index fcb4591a20..0000000000
--- a/sound/softsynth/fluidsynth.cpp
+++ /dev/null
@@ -1,254 +0,0 @@
-/* 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"
-
-#ifdef USE_FLUIDSYNTH
-
-#include "common/config-manager.h"
-#include "sound/musicplugin.h"
-#include "sound/mpu401.h"
-#include "sound/softsynth/emumidi.h"
-
-#include <fluidsynth.h>
-
-class MidiDriver_FluidSynth : public MidiDriver_Emulated {
-private:
-	MidiChannel_MPU401 _midiChannels[16];
-	fluid_settings_t *_settings;
-	fluid_synth_t *_synth;
-	int _soundFont;
-	int _outputRate;
-	Audio::SoundHandle _handle;
-
-protected:
-	// Because GCC complains about casting from const to non-const...
-	void setInt(const char *name, int val);
-	void setNum(const char *name, double num);
-	void setStr(const char *name, const char *str);
-
-	void generateSamples(int16 *buf, int len);
-
-public:
-	MidiDriver_FluidSynth(Audio::Mixer *mixer);
-
-	int open();
-	void close();
-	void send(uint32 b);
-
-	MidiChannel *allocateChannel();
-	MidiChannel *getPercussionChannel();
-
-	// AudioStream API
-	bool isStereo() const { return true; }
-	int getRate() const { return _outputRate; }
-};
-
-// MidiDriver method implementations
-
-MidiDriver_FluidSynth::MidiDriver_FluidSynth(Audio::Mixer *mixer)
-	: MidiDriver_Emulated(mixer) {
-
-	for (int i = 0; i < ARRAYSIZE(_midiChannels); i++) {
-		_midiChannels[i].init(this, i);
-	}
-
-	// It ought to be possible to get FluidSynth to generate samples at
-	// lower
-
-	_outputRate = _mixer->getOutputRate();
-	if (_outputRate < 22050)
-		_outputRate = 22050;
-	else if (_outputRate > 96000)
-		_outputRate = 96000;
-}
-
-void MidiDriver_FluidSynth::setInt(const char *name, int val) {
-	char *name2 = strdup(name);
-
-	fluid_settings_setint(_settings, name2, val);
-	free(name2);
-}
-
-void MidiDriver_FluidSynth::setNum(const char *name, double val) {
-	char *name2 = strdup(name);
-
-	fluid_settings_setnum(_settings, name2, val);
-	free(name2);
-}
-
-void MidiDriver_FluidSynth::setStr(const char *name, const char *val) {
-	char *name2 = strdup(name);
-	char *val2 = strdup(val);
-
-	fluid_settings_setstr(_settings, name2, val2);
-	free(name2);
-	free(val2);
-}
-
-int MidiDriver_FluidSynth::open() {
-	if (_isOpen)
-		return MERR_ALREADY_OPEN;
-
-	if (!ConfMan.hasKey("soundfont"))
-		error("FluidSynth requires a 'soundfont' setting");
-
-	_settings = new_fluid_settings();
-
-	// The default gain setting is ridiculously low - at least for me. This
-	// cannot be fixed by ScummVM's volume settings because they can only
-	// soften the sound, not amplify it, so instead we add an option to
-	// adjust the gain of FluidSynth itself.
-
-	double gain = (double)ConfMan.getInt("midi_gain") / 100.0;
-
-	setNum("synth.gain", gain);
-	setNum("synth.sample-rate", _outputRate);
-
-	_synth = new_fluid_synth(_settings);
-
-	// In theory, this ought to reduce CPU load... but it doesn't make any
-	// noticeable difference for me, so disable it for now.
-
-	// fluid_synth_set_interp_method(_synth, -1, FLUID_INTERP_LINEAR);
-	// fluid_synth_set_reverb_on(_synth, 0);
-	// fluid_synth_set_chorus_on(_synth, 0);
-
-	const char *soundfont = ConfMan.get("soundfont").c_str();
-
-	_soundFont = fluid_synth_sfload(_synth, soundfont, 1);
-	if (_soundFont == -1)
-		error("Failed loading custom sound font '%s'", soundfont);
-
-	MidiDriver_Emulated::open();
-
-	// The MT-32 emulator uses kSFXSoundType here. I don't know why.
-	_mixer->playStream(Audio::Mixer::kMusicSoundType, &_handle, this, -1, Audio::Mixer::kMaxChannelVolume, 0, DisposeAfterUse::NO, true);
-	return 0;
-}
-
-void MidiDriver_FluidSynth::close() {
-	if (!_isOpen)
-		return;
-	_isOpen = false;
-
-	_mixer->stopHandle(_handle);
-
-	if (_soundFont != -1)
-		fluid_synth_sfunload(_synth, _soundFont, 1);
-
-	delete_fluid_synth(_synth);
-	delete_fluid_settings(_settings);
-}
-
-void MidiDriver_FluidSynth::send(uint32 b) {
-	//byte param3 = (byte) ((b >> 24) & 0xFF);
-	uint param2 = (byte) ((b >> 16) & 0xFF);
-	uint param1 = (byte) ((b >>  8) & 0xFF);
-	byte cmd    = (byte) (b & 0xF0);
-	byte chan   = (byte) (b & 0x0F);
-
-	switch (cmd) {
-	case 0x80:	// Note Off
-		fluid_synth_noteoff(_synth, chan, param1);
-		break;
-	case 0x90:	// Note On
-		fluid_synth_noteon(_synth, chan, param1, param2);
-		break;
-	case 0xA0:	// Aftertouch
-		break;
-	case 0xB0:	// Control Change
-		fluid_synth_cc(_synth, chan, param1, param2);
-		break;
-	case 0xC0:	// Program Change
-		fluid_synth_program_change(_synth, chan, param1);
-		break;
-	case 0xD0:	// Channel Pressure
-		break;
-	case 0xE0:	// Pitch Bend
-		fluid_synth_pitch_bend(_synth, chan, (param2 << 7) | param1);
-		break;
-	case 0xF0:	// SysEx
-		// We should never get here! SysEx information has to be
-		// sent via high-level semantic methods.
-		warning("MidiDriver_FluidSynth: Receiving SysEx command on a send() call");
-		break;
-	default:
-		warning("MidiDriver_FluidSynth: Unknown send() command 0x%02X", cmd);
-		break;
-	}
-}
-
-MidiChannel *MidiDriver_FluidSynth::allocateChannel() {
-	for (int i = 0; i < ARRAYSIZE(_midiChannels); i++) {
-		if (i != 9 && _midiChannels[i].allocate())
-			return &_midiChannels[i];
-	}
-	return NULL;
-}
-
-MidiChannel *MidiDriver_FluidSynth::getPercussionChannel() {
-	return &_midiChannels[9];
-}
-
-void MidiDriver_FluidSynth::generateSamples(int16 *data, int len) {
-	fluid_synth_write_s16(_synth, len, data, 0, 2, data, 1, 2);
-}
-
-
-// Plugin interface
-
-class FluidSynthMusicPlugin : public MusicPluginObject {
-public:
-	const char *getName() const {
-		return "FluidSynth";
-	}
-
-	const char *getId() const {
-		return "fluidsynth";
-	}
-
-	MusicDevices getDevices() const;
-	Common::Error createInstance(MidiDriver **mididriver, MidiDriver::DeviceHandle = 0) const;
-};
-
-MusicDevices FluidSynthMusicPlugin::getDevices() const {
-	MusicDevices devices;
-	devices.push_back(MusicDevice(this, "", MT_GM));
-	return devices;
-}
-
-Common::Error FluidSynthMusicPlugin::createInstance(MidiDriver **mididriver, MidiDriver::DeviceHandle) const {
-	*mididriver = new MidiDriver_FluidSynth(g_system->getMixer());
-
-	return Common::kNoError;
-}
-
-//#if PLUGIN_ENABLED_DYNAMIC(FLUIDSYNTH)
-	//REGISTER_PLUGIN_DYNAMIC(FLUIDSYNTH, PLUGIN_TYPE_MUSIC, FluidSynthMusicPlugin);
-//#else
-	REGISTER_PLUGIN_STATIC(FLUIDSYNTH, PLUGIN_TYPE_MUSIC, FluidSynthMusicPlugin);
-//#endif
-
-#endif
diff --git a/sound/softsynth/fmtowns_pc98/towns_audio.cpp b/sound/softsynth/fmtowns_pc98/towns_audio.cpp
deleted file mode 100644
index 14bc840326..0000000000
--- a/sound/softsynth/fmtowns_pc98/towns_audio.cpp
+++ /dev/null
@@ -1,1583 +0,0 @@
-/* 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 "sound/softsynth/fmtowns_pc98/towns_audio.h"
-#include "common/endian.h"
-#include "backends/audiocd/audiocd.h"
-
-
-class TownsAudio_PcmChannel {
-friend class TownsAudioInterface;
-public:
-	TownsAudio_PcmChannel();
-	~TownsAudio_PcmChannel();
-
-private:
-	void loadExtData(uint8 *buffer, uint32 size);
-	void setupLoop(uint32 start, uint32 len);
-	void clear();
-
-	void envAttack();
-	void envDecay();
-	void envSustain();
-	void envRelease();
-
-	uint8 *curInstrument;
-	uint8 note;
-	uint8 velo;
-
-	int8 *data;
-	int8 *dataEnd;
-
-	int8 *loopEnd;
-	uint32 loopLen;
-
-	uint16 stepNote;
-	uint16 stepPitch;
-	uint16 step;
-
-	uint8 panLeft;
-	uint8 panRight;
-
-	uint32 pos;
-
-	uint8 envTotalLevel;
-	uint8 envAttackRate;
-	uint8 envDecayRate;
-	uint8 envSustainLevel;
-	uint8 envSustainRate;
-	uint8 envReleaseRate;
-
-	int16 envStep;
-	int16 envCurrentLevel;
-
-	EnvelopeState envState;
-
-	int8 *extData;
-};
-
-class TownsAudio_WaveTable {
-friend class TownsAudioInterface;
-public:
-	TownsAudio_WaveTable();
-	~TownsAudio_WaveTable();
-
-private:
-	void readHeader(const uint8 *buffer);
-	void readData(const uint8 *buffer);
-	void clear();
-
-	char name[9];
-	int32 id;
-	uint32 size;
-	uint32 loopStart;
-	uint32 loopLen;
-	uint16 rate;
-	uint16 rateOffs;
-	uint16 baseNote;
-	int8 *data;
-};
-
-TownsAudioInterface::TownsAudioInterface(Audio::Mixer *mixer, TownsAudioInterfacePluginDriver *driver) : TownsPC98_FmSynth(mixer, kTypeTowns),
-	_fmInstruments(0), _pcmInstruments(0), _pcmChan(0), _waveTables(0), _waveTablesTotalDataSize(0),
-	_baserate(55125.0f / (float)mixer->getOutputRate()), _tickLength(0), _timer(0), _drv(driver),
-	_pcmSfxChanMask(0),	_musicVolume(Audio::Mixer::kMaxMixerVolume), _sfxVolume(Audio::Mixer::kMaxMixerVolume),
-	_outputVolumeFlags(0), _outputMuteFlags(0), _pcmChanOut(0), _pcmChanReserved(0), _pcmChanKeyPressed(0),
-	_pcmChanEffectPlaying(0), _pcmChanKeyPlaying(0), _ready(false) {
-
-#define INTCB(x) &TownsAudioInterface::intf_##x
-	static const TownsAudioIntfCallback intfCb[] = {
-		// 0
-		INTCB(reset),
-		INTCB(keyOn),
-		INTCB(keyOff),
-		INTCB(setPanPos),
-		// 4
-		INTCB(setInstrument),
-		INTCB(loadInstrument),
-		INTCB(notImpl),
-		INTCB(setPitch),
-		// 8
-		INTCB(setLevel),
-		INTCB(chanOff),
-		INTCB(notImpl),
-		INTCB(notImpl),
-		// 12
-		INTCB(notImpl),
-		INTCB(notImpl),
-		INTCB(notImpl),
-		INTCB(notImpl),
-		// 16
-		INTCB(notImpl),
-		INTCB(writeReg),
-		INTCB(notImpl),
-		INTCB(writeRegBuffer),
-		// 20
-		INTCB(readRegBuffer),
-		INTCB(setTimerA),
-		INTCB(setTimerB),
-		INTCB(enableTimerA),
-		// 24
-		INTCB(enableTimerB),
-		INTCB(notImpl),
-		INTCB(notImpl),
-		INTCB(notImpl),
-		// 28
-		INTCB(notImpl),
-		INTCB(notImpl),
-		INTCB(notImpl),
-		INTCB(notImpl),
-		// 32
-		INTCB(loadSamples),
-		INTCB(reserveEffectChannels),
-		INTCB(loadWaveTable),
-		INTCB(unloadWaveTable),
-		// 36
-		INTCB(notImpl),
-		INTCB(pcmPlayEffect),
-		INTCB(notImpl),
-		INTCB(pcmChanOff),
-		// 40
-		INTCB(pcmEffectPlaying),
-		INTCB(notImpl),
-		INTCB(notImpl),
-		INTCB(notImpl),
-		// 44
-		INTCB(notImpl),
-		INTCB(notImpl),
-		INTCB(notImpl),
-		INTCB(notImpl),
-		// 48
-		INTCB(notImpl),
-		INTCB(notImpl),
-		INTCB(fmKeyOn),
-		INTCB(fmKeyOff),
-		// 52
-		INTCB(fmSetPanPos),
-		INTCB(fmSetInstrument),
-		INTCB(fmLoadInstrument),
-		INTCB(notImpl),
-		// 56
-		INTCB(fmSetPitch),
-		INTCB(fmSetLevel),
-		INTCB(fmReset),
-		INTCB(notImpl),
-		// 60
-		INTCB(notImpl),
-		INTCB(notImpl),
-		INTCB(notImpl),
-		INTCB(notImpl),
-		// 64
-		INTCB(notImpl),
-		INTCB(notImpl),
-		INTCB(notImpl),
-		INTCB(setOutputVolume),
-		// 68
-		INTCB(resetOutputVolume),
-		INTCB(notImpl),
-		INTCB(updateOutputVolume),
-		INTCB(notImpl),
-		// 72
-		INTCB(notImpl),
-		INTCB(cdaToggle),
-		INTCB(notImpl),
-		INTCB(notImpl),
-		// 76
-		INTCB(notImpl),
-		INTCB(notImpl),
-		INTCB(notImpl),
-		INTCB(notImpl),
-		// 80
-		INTCB(pcmUpdateEnvelopeGenerator),
-		INTCB(notImpl)
-	};
-#undef INTCB
-
-	_intfOpcodes = intfCb;
-
-	memset(_fmSaveReg, 0, sizeof(_fmSaveReg));
-	memset(_outputLevel, 0, sizeof(_outputLevel));
-
-	_timerBase = (uint32)(_baserate * 1000000.0f);
-	_tickLength = 2 * _timerBase;
-}
-
-TownsAudioInterface::~TownsAudioInterface() {
-	_ready = false;
-	deinit();
-
-	delete[] _fmSaveReg[0];
-	delete[] _fmSaveReg[1];
-	delete[] _fmInstruments;
-	delete[] _pcmInstruments;
-	delete[] _waveTables;
-	delete[] _pcmChan;
-}
-
-bool TownsAudioInterface::init() {
-	if (_ready)
-		return true;
-
-	if (!TownsPC98_FmSynth::init())
-		return false;
-
-	_fmSaveReg[0] = new uint8[256];
-	_fmSaveReg[1] = new uint8[256];
-	_fmInstruments = new uint8[128 * 48];
-	_pcmInstruments = new uint8[32 * 128];
-	_waveTables = new TownsAudio_WaveTable[128];
-	_pcmChan = new TownsAudio_PcmChannel[8];
-
-	_timer = 0;
-
-	setVolumeChannelMasks(-1, 0);
-
-	_ready = true;
-	callback(0);
-
-	return true;
-}
-
-int TownsAudioInterface::callback(int command, ...) {
-	if (!_ready)
-		return 1;
-
-	va_list args;
-	va_start(args, command);
-
-	if (command > 81) {
-		va_end(args);
-		return 4;
-	}
-
-	int res = (this->*_intfOpcodes[command])(args);
-
-	va_end(args);
-	return res;
-}
-
-void TownsAudioInterface::setMusicVolume(int volume) {
-	_musicVolume = CLIP<uint16>(volume, 0, Audio::Mixer::kMaxMixerVolume);
-	setVolumeIntern(_musicVolume, _sfxVolume);
-}
-
-void TownsAudioInterface::setSoundEffectVolume(int volume) {
-	_sfxVolume = CLIP<uint16>(volume, 0, Audio::Mixer::kMaxMixerVolume);
-	setVolumeIntern(_musicVolume, _sfxVolume);
-}
-
-void TownsAudioInterface::setSoundEffectChanMask(int mask) {
-	_pcmSfxChanMask = mask >> 6;
-	mask &= 0x3f;
-	setVolumeChannelMasks(~mask, mask);
-}
-
-void TownsAudioInterface::nextTickEx(int32 *buffer, uint32 bufferSize) {
-	if (!_ready)
-		return;
-
-	for (uint32 i = 0; i < bufferSize; i++) {
-		_timer += _tickLength;
-		while (_timer > 0x514767) {
-			_timer -= 0x514767;
-
-			for (int ii = 0; ii < 8; ii++) {
-				if ((_pcmChanKeyPlaying & _chanFlags[ii]) || (_pcmChanEffectPlaying & _chanFlags[ii])) {
-					TownsAudio_PcmChannel *s = &_pcmChan[ii];
-					s->pos += s->step;
-
-					if (&s->data[s->pos >> 11] >= s->loopEnd) {
-						if (s->loopLen) {
-							s->pos -= s->loopLen;
-						} else {
-							s->pos = 0;
-							_pcmChanEffectPlaying &= ~_chanFlags[ii];
-							_pcmChanKeyPlaying &= ~_chanFlags[ii];
-						}
-					}
-				}
-			}
-		}
-
-		int32 finOutL = 0;
-		int32 finOutR = 0;
-
-		for (int ii = 0; ii < 8; ii++) {
-			if (_pcmChanOut & _chanFlags[ii]) {
-				int32 o = _pcmChan[ii].data[_pcmChan[ii].pos >> 11] * _pcmChan[ii].velo;
-				if ((1 << ii) & (~_pcmSfxChanMask))
-					o = (o * _musicVolume) / Audio::Mixer::kMaxMixerVolume;
-				if ((1 << ii) & _pcmSfxChanMask)
-					o = (o * _sfxVolume) / Audio::Mixer::kMaxMixerVolume;
-				if (_pcmChan[ii].panLeft)
-					finOutL += ((o * _pcmChan[ii].panLeft) >> 3);
-				if (_pcmChan[ii].panRight)
-					finOutR += ((o * _pcmChan[ii].panRight) >> 3);
-				if (!((_pcmChanKeyPlaying & _chanFlags[ii]) || (_pcmChanEffectPlaying & _chanFlags[ii])))
-					_pcmChanOut &= ~_chanFlags[ii];
-			}
-		}
-
-		buffer[i << 1] += finOutL;
-		buffer[(i << 1) + 1] += finOutR;
-	}
-}
-
-void TownsAudioInterface::timerCallbackA() {
-	Common::StackLock lock(_mutex);
-	if (_drv && _ready)
-		_drv->timerCallback(0);
-}
-
-void TownsAudioInterface::timerCallbackB() {
-	Common::StackLock lock(_mutex);
-	if (_ready) {
-		if (_drv)
-			_drv->timerCallback(1);
-		callback(80);
-	}
-}
-
-int TownsAudioInterface::intf_reset(va_list &args) {
-	fmReset();
-	pcmReset();
-	callback(68);
-	return 0;
-}
-
-int TownsAudioInterface::intf_keyOn(va_list &args) {
-	int chan = va_arg(args, int);
-	int note = va_arg(args, int);
-	int velo = va_arg(args, int);
-	return (chan & 0x40) ? pcmKeyOn(chan, note, velo) : fmKeyOn(chan, note, velo);
-}
-
-int TownsAudioInterface::intf_keyOff(va_list &args) {
-	int chan = va_arg(args, int);
-	return (chan & 0x40) ? pcmKeyOff(chan) : fmKeyOff(chan);
-}
-
-int TownsAudioInterface::intf_setPanPos(va_list &args) {
-	int chan = va_arg(args, int);
-	int mode = va_arg(args, int);
-	return (chan & 0x40) ? pcmSetPanPos(chan, mode) : fmSetPanPos(chan, mode);
-}
-
-int TownsAudioInterface::intf_setInstrument(va_list &args) {
-	int chan = va_arg(args, int);
-	int instrId = va_arg(args, int);
-	return (chan & 0x40) ? pcmSetInstrument(chan, instrId) : fmSetInstrument(chan, instrId);
-}
-
-int TownsAudioInterface::intf_loadInstrument(va_list &args) {
-	int chanType = va_arg(args, int);
-	int instrId = va_arg(args, int);
-	uint8 *instrData = va_arg(args, uint8 *);
-	return (chanType & 0x40) ? pcmLoadInstrument(instrId, instrData) : fmLoadInstrument(instrId, instrData);
-}
-
-int TownsAudioInterface::intf_setPitch(va_list &args) {
-	int chan = va_arg(args, int);
-	int16 pitch = (int16)(va_arg(args, int) & 0xffff);
-	return (chan & 0x40) ? pcmSetPitch(chan, pitch) : fmSetPitch(chan, pitch);
-}
-
-int TownsAudioInterface::intf_setLevel(va_list &args) {
-	int chan = va_arg(args, int);
-	int lvl = va_arg(args, int);
-	return (chan & 0x40) ? pcmSetLevel(chan, lvl) : fmSetLevel(chan, lvl);
-}
-
-int TownsAudioInterface::intf_chanOff(va_list &args) {
-	int chan = va_arg(args, int);
-	return (chan & 0x40) ? pcmChanOff(chan) : fmChanOff(chan);
-}
-
-int TownsAudioInterface::intf_writeReg(va_list &args) {
-	int part = va_arg(args, int) ? 1 : 0;
-	int reg = va_arg(args, int);
-	int val = va_arg(args, int);
-	if ((!part && reg < 0x20) || (part && reg < 0x30) || (reg > 0xb6))
-		return 3;
-
-	bufferedWriteReg(part, reg, val);
-	return 0;
-}
-
-int TownsAudioInterface::intf_writeRegBuffer(va_list &args) {
-	int part = va_arg(args, int) ? 1 : 0;
-	int reg = va_arg(args, int);
-	int val = va_arg(args, int);
-
-	if ((!part && reg < 0x20) || (part && reg < 0x30) || (reg > 0xef))
-		return 3;
-
-	_fmSaveReg[part][reg] = val;
-	return 0;
-}
-
-int TownsAudioInterface::intf_readRegBuffer(va_list &args) {
-	int part = va_arg(args, int) ? 1 : 0;
-	int reg = va_arg(args, int);
-	uint8 *dst = va_arg(args, uint8 *);
-	*dst = 0;
-
-	if ((!part && reg < 0x20) || (part && reg < 0x30) || (reg > 0xef))
-		return 3;
-
-	*dst = _fmSaveReg[part][reg];
-	return 0;
-}
-
-int TownsAudioInterface::intf_setTimerA(va_list &args) {
-	int enable = va_arg(args, int);
-	int tempo = va_arg(args, int);
-
-	if (enable) {
-		bufferedWriteReg(0, 0x25, tempo & 3);
-		bufferedWriteReg(0, 0x24, (tempo >> 2) & 0xff);
-		bufferedWriteReg(0, 0x27, _fmSaveReg[0][0x27] | 0x05);
-	} else {
-		bufferedWriteReg(0, 0x27, (_fmSaveReg[0][0x27] & 0xfa) | 0x10);
-	}
-
-	return 0;
-}
-
-int TownsAudioInterface::intf_setTimerB(va_list &args) {
-	int enable = va_arg(args, int);
-	int tempo = va_arg(args, int);
-
-	if (enable) {
-		bufferedWriteReg(0, 0x26, tempo & 0xff);
-		bufferedWriteReg(0, 0x27, _fmSaveReg[0][0x27] | 0x0A);
-	} else {
-		bufferedWriteReg(0, 0x27, (_fmSaveReg[0][0x27] & 0xf5) | 0x20);
-	}
-
-	return 0;
-}
-
-int TownsAudioInterface::intf_enableTimerA(va_list &args) {
-	bufferedWriteReg(0, 0x27, _fmSaveReg[0][0x27] | 0x15);
-	return 0;
-}
-
-int TownsAudioInterface::intf_enableTimerB(va_list &args) {
-	bufferedWriteReg(0, 0x27, _fmSaveReg[0][0x27] | 0x2a);
-	return 0;
-}
-
-int TownsAudioInterface::intf_loadSamples(va_list &args) {
-	uint32 dest = va_arg(args, uint32);
-	int size = va_arg(args, int);
-	uint8 *src = va_arg(args, uint8*);
-
-	if (dest >= 65536 || size == 0 || size > 65536)
-		return 3;
-	if (size + dest > 65536)
-		return 5;
-
-	int dwIndex = _numWaveTables - 1;
-	for (uint32 t = _waveTablesTotalDataSize; dwIndex && (dest < t); dwIndex--)
-		t -= _waveTables[dwIndex].size;
-
-	TownsAudio_WaveTable *s = &_waveTables[dwIndex];
-	_waveTablesTotalDataSize -= s->size;
-	s->size = size;
-	s->readData(src);
-	_waveTablesTotalDataSize += s->size;
-
-	return 0;
-}
-
-int TownsAudioInterface::intf_reserveEffectChannels(va_list &args) {
-	int numChan = va_arg(args, int);
-	if (numChan > 8)
-		return 3;
-	if ((numChan << 13) + _waveTablesTotalDataSize > 65536)
-		return 5;
-
-	if (numChan == _numReservedChannels)
-		return 0;
-
-	if (numChan < _numReservedChannels) {
-		int c = 8 - _numReservedChannels;
-		for (int i = numChan; i; i--) {
-			uint8 f = ~_chanFlags[c--];
-			_pcmChanEffectPlaying &= f;
-		}
-	} else {
-		int c = 7 - _numReservedChannels;
-		for (int i = numChan - _numReservedChannels; i; i--) {
-			uint8 f = ~_chanFlags[c--];
-			_pcmChanKeyPressed &= f;
-			_pcmChanKeyPlaying &= f;
-		}
-	}
-
-	static const uint8 reserveChanFlags[] = { 0x00, 0x80, 0xC0, 0xE0, 0xF0, 0xF8, 0xFC, 0xFE, 0xFF };
-	_numReservedChannels = numChan;
-	_pcmChanReserved = reserveChanFlags[_numReservedChannels];
-
-	return 0;
-}
-
-int TownsAudioInterface::intf_loadWaveTable(va_list &args) {
-	uint8 *data = va_arg(args, uint8 *);
-	if (_numWaveTables > 127)
-		return 3;
-
-	TownsAudio_WaveTable w;
-	w.readHeader(data);
-	if (!w.size)
-		return 6;
-
-	if (_waveTablesTotalDataSize + w.size > 65504)
-		return 5;
-
-	for (int i = 0; i < _numWaveTables; i++) {
-		if (_waveTables[i].id == w.id)
-			return 10;
-	}
-
-	TownsAudio_WaveTable *s = &_waveTables[_numWaveTables++];
-	s->readHeader(data);
-
-	_waveTablesTotalDataSize += s->size;
-	callback(32, _waveTablesTotalDataSize, s->size, data + 32);
-
-	return 0;
-}
-
-int TownsAudioInterface::intf_unloadWaveTable(va_list &args) {
-	int id = va_arg(args, int);
-
-	if (id == -1) {
-		for (int i = 0; i < 128; i++)
-			_waveTables[i].clear();
-		_numWaveTables = 0;
-		_waveTablesTotalDataSize = 0;
-	} else {
-		if (_waveTables) {
-			for (int i = 0; i < _numWaveTables; i++) {
-				if (_waveTables[i].id == id) {
-					_numWaveTables--;
-					_waveTablesTotalDataSize -= _waveTables[i].size;
-					_waveTables[i].clear();
-					for (; i < _numWaveTables; i++)
-						memcpy(&_waveTables[i], &_waveTables[i + 1], sizeof(TownsAudio_WaveTable));
-					return 0;
-				}
-				return 9;
-			}
-		}
-	}
-
-	return 0;
-}
-
-int TownsAudioInterface::intf_pcmPlayEffect(va_list &args) {
-	int chan = va_arg(args, int);
-	int note = va_arg(args, int);
-	int velo = va_arg(args, int);
-	uint8 *data = va_arg(args, uint8 *);
-
-	if (chan < 0x40 || chan > 0x47)
-		return 1;
-
-	if (note & 0x80 || velo & 0x80)
-		return 3;
-
-	chan -= 0x40;
-
-	if (!(_pcmChanReserved & _chanFlags[chan]))
-		return 7;
-
-	if ((_pcmChanEffectPlaying & _chanFlags[chan]))
-		return 2;
-
-	TownsAudio_WaveTable w;
-	w.readHeader(data);
-
-	if (w.size < (w.loopStart + w.loopLen))
-		return 13;
-
-	if (!w.size)
-		return 6;
-
-	TownsAudio_PcmChannel *p = &_pcmChan[chan];
-
-	_pcmChanNote[chan] = note;
-	_pcmChanVelo[chan] = velo;
-
-	p->note = note;
-	p->velo = velo << 1;
-
-	p->loadExtData(data + 32, w.size);
-	p->setupLoop(w.loopStart, w.loopLen);
-
-	pcmCalcPhaseStep(p, &w);
-	if (p->step > 2048)
-		p->step = 2048;
-
-	_pcmChanEffectPlaying |= _chanFlags[chan];
-	_pcmChanOut |= _chanFlags[chan];
-
-	return 0;
-}
-
-int TownsAudioInterface::intf_pcmChanOff(va_list &args) {
-	int chan = va_arg(args, int);
-	pcmChanOff(chan);
-	return 0;
-}
-
-int TownsAudioInterface::intf_pcmEffectPlaying(va_list &args) {
-	int chan = va_arg(args, int);
-	if (chan < 0x40 || chan > 0x47)
-		return 1;
-	chan -= 0x40;
-	return (_pcmChanEffectPlaying & _chanFlags[chan]) ? 1 : 0;
-}
-
-int TownsAudioInterface::intf_fmKeyOn(va_list &args) {
-	int chan = va_arg(args, int);
-	int note = va_arg(args, int);
-	int velo = va_arg(args, int);
-	return fmKeyOn(chan, note, velo);
-}
-
-int TownsAudioInterface::intf_fmKeyOff(va_list &args) {
-	int chan = va_arg(args, int);
-	return fmKeyOff(chan);
-}
-
-int TownsAudioInterface::intf_fmSetPanPos(va_list &args) {
-	int chan = va_arg(args, int);
-	int mode = va_arg(args, int);
-	return fmSetPanPos(chan, mode);
-}
-
-int TownsAudioInterface::intf_fmSetInstrument(va_list &args) {
-	int chan = va_arg(args, int);
-	int instrId = va_arg(args, int);
-	return fmSetInstrument(chan, instrId);
-}
-
-int TownsAudioInterface::intf_fmLoadInstrument(va_list &args) {
-	int instrId = va_arg(args, int);
-	uint8 *instrData = va_arg(args, uint8 *);
-	return fmLoadInstrument(instrId, instrData);
-}
-
-int TownsAudioInterface::intf_fmSetPitch(va_list &args) {
-	int chan = va_arg(args, int);
-	uint16 freq = va_arg(args, int) & 0xffff;
-	return fmSetPitch(chan, freq);
-}
-
-int TownsAudioInterface::intf_fmSetLevel(va_list &args) {
-	int chan = va_arg(args, int);
-	int lvl = va_arg(args, int);
-	return fmSetLevel(chan, lvl);
-}
-
-int TownsAudioInterface::intf_fmReset(va_list &args) {
-	fmReset();
-	return 0;
-}
-
-int TownsAudioInterface::intf_setOutputVolume(va_list &args) {
-	int chanType = va_arg(args, int);
-	int left = va_arg(args, int);
-	int right = va_arg(args, int);
-
-	if (left & 0xff80 || right & 0xff80)
-		return 3;
-
-	static const uint8 flags[] = { 0x0C, 0x30, 0x40, 0x80 };
-
-	uint8 chan = (chanType & 0x40) ? 8 : 12;
-
-	chanType &= 3;
-	left = (left & 0x7e) >> 1;
-	right = (right & 0x7e) >> 1;
-
-	if (chan)
-		_outputVolumeFlags |= flags[chanType];
-	else
-		_outputVolumeFlags &= ~flags[chanType];
-
-	if (chanType > 1) {
-		_outputLevel[chan + chanType] = left;
-	} else {
-		if (chanType == 0)
-			chan -= 8;
-		_outputLevel[chan] = left;
-		_outputLevel[chan + 1] = right;
-	}
-
-	updateOutputVolume();
-
-	return 0;
-}
-
-int TownsAudioInterface::intf_resetOutputVolume(va_list &args) {
-	memset(_outputLevel, 0, sizeof(_outputLevel));
-	_outputMuteFlags = 0;
-	_outputVolumeFlags = 0;
-	updateOutputVolume();
-	return 0;
-}
-
-int TownsAudioInterface::intf_updateOutputVolume(va_list &args) {
-	int flags = va_arg(args, int);
-	_outputMuteFlags = flags & 3;
-	updateOutputVolume();
-	return 0;
-}
-
-int TownsAudioInterface::intf_cdaToggle(va_list &args) {
-	//int mode = va_arg(args, int);
-	//_unkMask = mode ? 0x7f : 0x3f;
-	return 0;
-}
-
-int TownsAudioInterface::intf_pcmUpdateEnvelopeGenerator(va_list &args) {
-	for (int i = 0; i < 8; i++)
-		pcmUpdateEnvelopeGenerator(i);
-	return 0;
-}
-
-int TownsAudioInterface::intf_notImpl(va_list &args) {
-	return 4;
-}
-
-void TownsAudioInterface::fmReset() {
-	TownsPC98_FmSynth::reset();
-
-	_fmChanPlaying = 0;
-	memset(_fmChanNote, 0, sizeof(_fmChanNote));
-	memset(_fmChanPitch, 0, sizeof(_fmChanPitch));
-
-	memset(_fmSaveReg[0], 0, 240);
-	memset(&_fmSaveReg[0][240], 0x7f, 16);
-	memset(_fmSaveReg[1], 0, 256);
-	memset(&_fmSaveReg[1][240], 0x7f, 16);
-	_fmSaveReg[0][243] = _fmSaveReg[0][247] = _fmSaveReg[0][251] = _fmSaveReg[0][255] = _fmSaveReg[1][243] = _fmSaveReg[1][247] = _fmSaveReg[1][251] = _fmSaveReg[1][255] = 0xff;
-
-	for (int i = 0; i < 128; i++)
-		fmLoadInstrument(i, _fmDefaultInstrument);
-
-	bufferedWriteReg(0, 0x21, 0);
-	bufferedWriteReg(0, 0x2C, 0x80);
-	bufferedWriteReg(0, 0x2B, 0);
-	bufferedWriteReg(0, 0x27, 0x30);
-
-	for (int i = 0; i < 6; i++) {
-		fmKeyOff(i);
-		fmSetInstrument(i, 0);
-		fmSetLevel(i, 127);
-	}
-}
-
-int TownsAudioInterface::fmKeyOn(int chan, int note, int velo) {
-	if (chan > 5)
-		return 1;
-	if (note < 12 || note > 107 || (velo & 0x80))
-		return 3;
-	if (_fmChanPlaying & _chanFlags[chan])
-		return 2;
-
-	_fmChanPlaying |= _chanFlags[chan];
-	note -= 12;
-
-	_fmChanNote[chan] = note;
-	int16 pitch = _fmChanPitch[chan];
-
-	uint8 part = chan > 2 ? 1 : 0;
-	if (chan > 2)
-		chan -= 3;
-
-	int frq = 0;
-	uint8 bl = 0;
-
-	if (note) {
-		frq = _frequency[(note - 1) % 12];
-		bl = (note - 1) / 12;
-	} else {
-		frq = 616;
-	}
-
-	frq += pitch;
-
-	if (frq < 616) {
-		if (!bl) {
-			frq = 616;
-		} else {
-			frq += 616;
-			--bl;
-		}
-	} else if (frq > 1232) {
-		if (bl == 7) {
-			frq = 15500;
-		} else {
-			frq -= 616;
-			++bl;
-		}
-	}
-
-	frq |= (bl << 11);
-
-	bufferedWriteReg(part, chan + 0xa4, (frq >> 8) & 0xff);
-	bufferedWriteReg(part, chan + 0xa0, frq & 0xff);
-
-	velo = (velo >> 2) + 96;
-	uint16 c = _carrier[_fmSaveReg[part][0xb0 + chan] & 7];
-	_fmSaveReg[part][0xe0 + chan] = velo;
-
-	for (uint8 reg = 0x40 + chan; reg < 0x50; reg += 4) {
-		c += c;
-		if (c & 0x100) {
-			c &= 0xff;
-			bufferedWriteReg(part, reg, (((((((_fmSaveReg[part][0x80 + reg] ^ 0x7f) * velo) >> 7) + 1) * _fmSaveReg[part][0xd0 + chan]) >> 7) + 1) ^ 0x7f);
-		}
-	}
-
-	uint8 v = chan;
-	if (part)
-		v |= 4;
-
-	for (uint8 reg = 0x80 + chan; reg < 0x90; reg += 4)
-		writeReg(part, reg, _fmSaveReg[part][reg] | 0x0f);
-
-	writeReg(0, 0x28, v);
-
-	for (uint8 reg = 0x80 + chan; reg < 0x90; reg += 4)
-		writeReg(part, reg, _fmSaveReg[part][reg]);
-
-	bufferedWriteReg(0, 0x28, v | 0xf0);
-
-	return 0;
-}
-
-int TownsAudioInterface::fmKeyOff(int chan) {
-	if (chan > 5)
-		return 1;
-	_fmChanPlaying &= ~_chanFlags[chan];
-	if (chan > 2)
-		chan++;
-	bufferedWriteReg(0, 0x28, chan);
-	return 0;
-}
-
-int TownsAudioInterface::fmChanOff(int chan) {
-	if (chan > 5)
-		return 1;
-	_fmChanPlaying &= ~_chanFlags[chan];
-
-	uint8 part = chan > 2 ? 1 : 0;
-	if (chan > 2)
-		chan -= 3;
-
-	for (uint8 reg = 0x80 + chan; reg < 0x90; reg += 4)
-		writeReg(part, reg, _fmSaveReg[part][reg] | 0x0f);
-
-	if (part)
-		chan += 4;
-	writeReg(0, 0x28, chan);
-	return 0;
-}
-
-int TownsAudioInterface::fmSetPanPos(int chan, int value) {
-	if (chan > 5)
-		return 1;
-
-	uint8 part = chan > 2 ? 1 : 0;
-	if (chan > 2)
-		chan -= 3;
-
-	if (value > 0x40)
-		value = 0x40;
-	else if (value < 0x40)
-		value = 0x80;
-	else
-		value = 0xC0;
-
-	bufferedWriteReg(part, 0xb4 + chan, (_fmSaveReg[part][0xb4 + chan] & 0x3f) | value);
-	return 0;
-}
-
-int TownsAudioInterface::fmSetInstrument(int chan, int instrId) {
-	if (chan > 5)
-		return 1;
-	if (instrId > 127)
-		return 3;
-
-	uint8 part = chan > 2 ? 1 : 0;
-	if (chan > 2)
-		chan -= 3;
-
-	uint8 *src = &_fmInstruments[instrId * 48 + 8];
-
-	uint16 c = _carrier[src[24] & 7];
-	uint8 reg = 0x30 + chan;
-
-	for (; reg < 0x40; reg += 4)
-		bufferedWriteReg(part, reg, *src++);
-
-	for (; reg < 0x50; reg += 4) {
-		uint8 v = *src++;
-		_fmSaveReg[part][0x80 + reg] = _fmSaveReg[part][reg] = v;
-		c += c;
-		if (c & 0x100) {
-			c &= 0xff;
-			v = 127;
-		}
-		writeReg(part, reg, v);
-	}
-
-	for (; reg < 0x90; reg += 4)
-		bufferedWriteReg(part, reg, *src++);
-
-	reg += 0x20;
-	bufferedWriteReg(part, reg, *src++);
-
-	uint8 v = *src++;
-	reg += 4;
-	if (v < 64)
-		v |= (_fmSaveReg[part][reg] & 0xc0);
-	bufferedWriteReg(part, reg, v);
-
-	return 0;
-}
-
-int TownsAudioInterface::fmLoadInstrument(int instrId, const uint8 *data) {
-	if (instrId > 127)
-		return 3;
-	assert(data);
-	memcpy(&_fmInstruments[instrId * 48], data, 48);
-	return 0;
-}
-
-int TownsAudioInterface::fmSetPitch(int chan, int pitch) {
-	if (chan > 5)
-		return 1;
-
-	uint8 bl = _fmChanNote[chan];
-	int frq = 0;
-
-	if (pitch < 0) {
-		if (bl) {
-			if (pitch < -8008)
-				pitch = -8008;
-			pitch *= -1;
-			pitch /= 13;
-			frq = _frequency[(bl - 1) % 12] - pitch;
-			bl = (bl - 1) / 12;
-			_fmChanPitch[chan] = -pitch;
-
-			if (frq < 616) {
-				if (bl) {
-					frq += 616;
-					bl--;
-				} else {
-					frq = 616;
-					bl = 0;
-				}
-			}
-		} else {
-			frq = 616;
-			bl = 0;
-		}
-
-	} else if (pitch > 0) {
-		if (bl < 96) {
-			if (pitch > 8008)
-				pitch = 8008;
-			pitch /= 13;
-
-			if (bl) {
-				frq = _frequency[(bl - 1) % 12] + pitch;
-				bl = (bl - 1) / 12;
-			} else {
-				frq = 616;
-				bl = 0;
-			}
-
-			_fmChanPitch[chan] = pitch;
-
-			if (frq > 1232) {
-				if (bl < 7) {
-					frq -= 616;
-					bl++;
-				} else {
-					frq = 1164;
-					bl = 7;
-				}
-			} else {
-				if (bl >= 7 && frq > 1164)
-					frq = 1164;
-			}
-
-		} else {
-			frq = 1164;
-			bl = 7;
-		}
-	} else {
-		_fmChanPitch[chan] = 0;
-		if (bl) {
-			frq = _frequency[(bl - 1) % 12];
-			bl = (bl - 1) / 12;
-		} else {
-			frq = 616;
-			bl = 0;
-		}
-	}
-
-	uint8 part = chan > 2 ? 1 : 0;
-	if (chan > 2)
-		chan -= 3;
-
-	frq |= (bl << 11);
-
-	bufferedWriteReg(part, chan + 0xa4, (frq >> 8));
-	bufferedWriteReg(part, chan + 0xa0, (frq & 0xff));
-
-	return 0;
-}
-
-int TownsAudioInterface::fmSetLevel(int chan, int lvl) {
-	if (chan > 5)
-		return 1;
-	if (lvl > 127)
-		return 3;
-
-	uint8 part = chan > 2 ? 1 : 0;
-	if (chan > 2)
-		chan -= 3;
-
-	uint16 c = _carrier[_fmSaveReg[part][0xb0 + chan] & 7];
-	_fmSaveReg[part][0xd0 + chan] = lvl;
-
-	for (uint8 reg = 0x40 + chan; reg < 0x50; reg += 4) {
-		c += c;
-		if (c & 0x100) {
-			c &= 0xff;
-			bufferedWriteReg(part, reg, (((((((_fmSaveReg[part][0x80 + reg] ^ 0x7f) * lvl) >> 7) + 1) * _fmSaveReg[part][0xe0 + chan]) >> 7) + 1) ^ 0x7f);
-		}
-	}
-	return 0;
-}
-
-void TownsAudioInterface::bufferedWriteReg(uint8 part, uint8 regAddress, uint8 value) {
-	_fmSaveReg[part][regAddress] = value;
-	writeReg(part, regAddress, value);
-}
-
-void TownsAudioInterface::pcmReset() {
-	_pcmChanOut = 0;
-	_pcmChanReserved = _pcmChanKeyPressed = _pcmChanEffectPlaying = _pcmChanKeyPlaying = 0;
-	_numReservedChannels = 0;
-
-	memset(_pcmChanNote, 0, 8);
-	memset(_pcmChanVelo, 0, 8);
-	memset(_pcmChanLevel, 0, 8);
-
-	for (int i = 0; i < 8; i++)
-		_pcmChan[i].clear();
-
-	memset(_pcmInstruments, 0, 128 * 32);
-	static uint8 name[] = { 0x4E, 0x6F, 0x20, 0x44, 0x61, 0x74, 0x61, 0x21 };
-	for (int i = 0; i < 32; i++)
-		memcpy(_pcmInstruments + i * 128, name, 8);
-
-	for (int i = 0; i < 128; i++)
-		_waveTables[i].clear();
-	_numWaveTables = 0;
-	_waveTablesTotalDataSize = 0;
-
-	for (int i = 0x40; i < 0x48; i++) {
-		pcmSetInstrument(i, 0);
-		pcmSetLevel(i, 127);
-	}
-}
-
-int TownsAudioInterface::pcmKeyOn(int chan, int note, int velo) {
-	if (chan < 0x40 || chan > 0x47)
-		return 1;
-
-	if (note & 0x80 || velo & 0x80)
-		return 3;
-
-	chan -= 0x40;
-
-	if ((_pcmChanReserved & _chanFlags[chan]) || (_pcmChanKeyPressed & _chanFlags[chan]))
-		return 2;
-
-	_pcmChanNote[chan] = note;
-	_pcmChanVelo[chan] = velo;
-
-	TownsAudio_PcmChannel *p = &_pcmChan[chan];
-	p->note = note;
-
-	uint8 *instr = _pcmChan[chan].curInstrument;
-	int i = 0;
-	for (; i < 8; i++) {
-		if (note <= instr[16 + 2 * i])
-			break;
-	}
-
-	if (i == 8)
-		return 8;
-
-	int il = i << 3;
-	p->note += instr[il + 70];
-
-	p->envTotalLevel = instr[il + 64];
-	p->envAttackRate = instr[il + 65];
-	p->envDecayRate = instr[il + 66];
-	p->envSustainLevel = instr[il + 67];
-	p->envSustainRate = instr[il + 68];
-	p->envReleaseRate = instr[il + 69];
-	p->envStep = 0;
-
-	int32 id = (int32)READ_LE_UINT32(&instr[i * 4 + 32]);
-
-	for (i = 0; i < _numWaveTables; i++) {
-		if (id == _waveTables[i].id)
-			break;
-	}
-
-	if (i == _numWaveTables)
-		return 9;
-
-	TownsAudio_WaveTable *w = &_waveTables[i];
-
-	p->data = w->data;
-	p->dataEnd = w->data + w->size;
-	p->setupLoop(w->loopStart, w->loopLen);
-
-	pcmCalcPhaseStep(p, w);
-
-	uint32 lvl = _pcmChanLevel[chan] * _pcmChanVelo[chan];
-	p->envTotalLevel = ((p->envTotalLevel * lvl) >> 14) & 0xff;
-	p->envSustainLevel = ((p->envSustainLevel * lvl) >> 14) & 0xff;
-
-	p->envAttack();
-	p->velo = (p->envCurrentLevel >> 8) << 1;
-
-	_pcmChanKeyPressed |= _chanFlags[chan];
-	_pcmChanKeyPlaying |= _chanFlags[chan];
-	_pcmChanOut |= _chanFlags[chan];
-
-	return 0;
-}
-
-int TownsAudioInterface::pcmKeyOff(int chan) {
-	if (chan < 0x40 || chan > 0x47)
-		return 1;
-
-	chan -= 0x40;
-	_pcmChanKeyPressed &= ~_chanFlags[chan];
-	_pcmChan[chan].envRelease();
-	return 0;
-}
-
-int TownsAudioInterface::pcmChanOff(int chan) {
-	if (chan < 0x40 || chan > 0x47)
-		return 1;
-
-	chan -= 0x40;
-
-	_pcmChanKeyPressed &= ~_chanFlags[chan];
-	_pcmChanEffectPlaying &= ~_chanFlags[chan];
-	_pcmChanKeyPlaying &= ~_chanFlags[chan];
-	_pcmChanOut &= ~_chanFlags[chan];
-
-	return 0;
-}
-
-int TownsAudioInterface::pcmSetPanPos(int chan, int mode) {
-	if (chan > 0x47)
-		return 1;
-	if (mode & 0x80)
-		return 3;
-
-	chan -= 0x40;
-	uint8 blc = 0x77;
-
-	if (mode > 64) {
-		mode -= 64;
-		blc = ((blc ^ (mode >> 3)) + (mode << 4)) & 0xff;
-	} else if (mode < 64) {
-		mode = (mode >> 3) ^ 7;
-		blc = ((119 + mode) ^ (mode << 4)) & 0xff;
-	}
-
-	_pcmChan[chan].panLeft = blc & 0x0f;
-	_pcmChan[chan].panRight = blc >> 4;
-
-	return 0;
-}
-
-int TownsAudioInterface::pcmSetInstrument(int chan, int instrId) {
-	if (chan > 0x47)
-		return 1;
-	if (instrId > 31)
-		return 3;
-	chan -= 0x40;
-	_pcmChan[chan].curInstrument = &_pcmInstruments[instrId * 128];
-	return 0;
-}
-
-int TownsAudioInterface::pcmLoadInstrument(int instrId, const uint8 *data) {
-	if (instrId > 31)
-		return 3;
-	assert(data);
-	memcpy(&_pcmInstruments[instrId * 128], data, 128);
-	return 0;
-}
-
-int TownsAudioInterface::pcmSetPitch(int chan, int pitch) {
-	if (chan > 0x47)
-		return 1;
-
-	if (pitch < -8192 || pitch > 8191)
-		return 3;
-
-	chan -= 0x40;
-	TownsAudio_PcmChannel *p = &_pcmChan[chan];
-
-	uint32 pts = 0x4000;
-
-	if (pitch < 0)
-		pts = (0x20000000 / (-pitch + 0x2001)) >> 2;
-	else if (pitch > 0)
-		pts = (((pitch + 0x2001) << 16) / 0x2000) >> 2;
-
-	p->stepPitch = pts & 0xffff;
-	p->step = (p->stepNote * p->stepPitch) >> 14;
-
-//	if (_pcmChanUnkFlag & _chanFlags[chan])
-//		 unk[chan] = (((p->step * 1000) << 11) / 98) / 20833;
-
-	/*else*/
-	if ((_pcmChanEffectPlaying & _chanFlags[chan]) && (p->step > 2048))
-		p->step = 2048;
-
-	return 0;
-}
-
-int TownsAudioInterface::pcmSetLevel(int chan, int lvl) {
-	if (chan > 0x47)
-		return 1;
-
-	if (lvl & 0x80)
-		return 3;
-
-	chan -= 0x40;
-	TownsAudio_PcmChannel *p = &_pcmChan[chan];
-
-	if (_pcmChanReserved & _chanFlags[chan]) {
-		_pcmChanVelo[chan] = lvl;
-		p->velo = lvl << 1;
-	} else {
-		int32 t = p->envStep * lvl;
-		if (_pcmChanLevel[chan])
-			t /= _pcmChanLevel[chan];
-		p->envStep = t;
-		t = p->envCurrentLevel * lvl;
-		if (_pcmChanLevel[chan])
-			t /= _pcmChanLevel[chan];
-		p->envCurrentLevel = t;
-		_pcmChanLevel[chan] = lvl;
-		p->velo = p->envCurrentLevel >> 8;
-	}
-
-	return 0;
-}
-
-void TownsAudioInterface::pcmUpdateEnvelopeGenerator(int chan) {
-	TownsAudio_PcmChannel *p = &_pcmChan[chan];
-	if (!p->envCurrentLevel) {
-		_pcmChanKeyPlaying &= ~_chanFlags[chan];
-		p->envState = kEnvReady;
-	}
-
-	if (!(_pcmChanKeyPlaying & _chanFlags[chan]))
-		return;
-
-	switch (p->envState) {
-	case kEnvAttacking:
-		if (((p->envCurrentLevel + p->envStep) >> 8) > p->envTotalLevel) {
-			p->envDecay();
-			return;
-		} else {
-			p->envCurrentLevel += p->envStep;
-		}
-		break;
-
-	case kEnvDecaying:
-		if (((p->envCurrentLevel - p->envStep) >> 8) < p->envSustainLevel) {
-			p->envSustain();
-			return;
-		} else {
-			p->envCurrentLevel -= p->envStep;
-		}
-		break;
-
-	case kEnvSustaining:
-	case kEnvReleasing:
-		p->envCurrentLevel -= p->envStep;
-		if (p->envCurrentLevel <= 0)
-			p->envCurrentLevel = 0;
-		break;
-
-	default:
-		break;
-	}
-	p->velo = (p->envCurrentLevel >> 8) << 1;
-}
-
-void TownsAudioInterface::pcmCalcPhaseStep(TownsAudio_PcmChannel *p, TownsAudio_WaveTable *w) {
-	int8 diff = p->note - w->baseNote;
-	uint16 r = w->rate + w->rateOffs;
-	uint16 bl = 0;
-	uint32 s = 0;
-
-	if (diff < 0) {
-		diff *= -1;
-		bl = diff % 12;
-		diff /= 12;
-		s = (r >> diff);
-		if (bl)
-			s = (s * _pcmPhase2[bl]) >> 16;
-
-	} else if (diff > 0) {
-		bl = diff % 12;
-		diff /= 12;
-		s = (r << diff);
-		if (bl)
-			s += ((s * _pcmPhase1[bl]) >> 16);
-
-	} else {
-		s = r;
-	}
-
-	p->stepNote = s & 0xffff;
-	p->step = (s * p->stepPitch) >> 14;
-}
-
-void TownsAudioInterface::updateOutputVolume() {
-	// FM Towns seems to support volumes of 0 - 63 for each channel.
-	// We recalculate sane values for our 0 to 255 volume range and
-	// balance values for our -128 to 127 volume range
-
-	// CD-AUDIO
-	uint32 maxVol = MAX(_outputLevel[12], _outputLevel[13]);
-
-	int volume = (int)(((float)(maxVol * 255) / 63.0f));
-	int balance = maxVol ? (int)( ( ((int)_outputLevel[13] - _outputLevel[12]) * 127) / (float)maxVol) : 0;
-
-	g_system->getAudioCDManager()->setVolume(volume);
-	g_system->getAudioCDManager()->setBalance(balance);
-}
-
-const uint8 TownsAudioInterface::_chanFlags[] = {
-	0x01, 0x02, 0x04, 0x08, 0x10, 0x20, 0x40, 0x80
-};
-
-const uint16 TownsAudioInterface::_frequency[] = {
-	0x028C, 0x02B4, 0x02DC, 0x030A, 0x0338, 0x0368, 0x039C, 0x03D4, 0x040E, 0x044A, 0x048C, 0x04D0
-};
-
-const uint8 TownsAudioInterface::_carrier[] = {
-	0x10, 0x10, 0x10, 0x10, 0x30, 0x70, 0x70, 0xF0
-};
-
-const uint8 TownsAudioInterface::_fmDefaultInstrument[] = {
-	0x45, 0x4C, 0x45, 0x50, 0x49, 0x41, 0x4E, 0x4F, 0x01, 0x0A, 0x02, 0x01,
-	0x1E, 0x32, 0x05, 0x00, 0x9C, 0xDC, 0x9C, 0xDC, 0x07, 0x03, 0x14, 0x08,
-	0x00, 0x03, 0x05, 0x05, 0x55, 0x45, 0x27, 0xA7, 0x04, 0xC0, 0x00, 0x00,
-	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
-};
-
-const uint16 TownsAudioInterface::_pcmPhase1[] =  {
-	0x879B, 0x0F37, 0x1F58, 0x306E, 0x4288, 0x55B6, 0x6A08, 0x7F8F, 0x965E, 0xAE88, 0xC882, 0xE341
-};
-
-const uint16 TownsAudioInterface::_pcmPhase2[] =  {
-	0xFEFE, 0xF1A0, 0xE411, 0xD744, 0xCB2F, 0xBFC7, 0xB504, 0xAAE2, 0xA144, 0x9827, 0x8FAC
-};
-
-TownsAudio_PcmChannel::TownsAudio_PcmChannel() {
-	extData = 0;
-	clear();
-}
-
-TownsAudio_PcmChannel::~TownsAudio_PcmChannel() {
-	clear();
-}
-
-void TownsAudio_PcmChannel::loadExtData(uint8 *buffer, uint32 size) {
-	delete[] extData;
-	extData = new int8[size];
-	int8 *src = (int8 *)buffer;
-	int8 *dst = extData;
-	for (uint32 i = 0; i < size; i++)
-		*dst++ = *src & 0x80 ? (*src++ & 0x7f) : -*src++;
-
-	data = extData;
-	dataEnd = extData + size;
-	pos = 0;
-}
-
-void TownsAudio_PcmChannel::setupLoop(uint32 start, uint32 len) {
-	loopLen = len << 11;
-	loopEnd = loopLen ? &data[(start + loopLen) >> 11] : dataEnd;
-	pos = start;
-}
-
-void TownsAudio_PcmChannel::clear() {
-	curInstrument = 0;
-	note = 0;
-	velo = 0;
-
-	data = 0;
-	dataEnd = 0;
-	loopLen = 0;
-
-	pos = 0;
-	loopEnd = 0;
-
-	step = 0;
-	stepNote = 0x4000;
-	stepPitch = 0x4000;
-
-	panLeft = panRight = 7;
-
-	envTotalLevel = envAttackRate = envDecayRate = envSustainLevel = envSustainRate = envReleaseRate = 0;
-	envStep = envCurrentLevel = 0;
-
-	envState = kEnvReady;
-
-	delete[] extData;
-	extData = 0;
-}
-
-void TownsAudio_PcmChannel::envAttack() {
-	envState = kEnvAttacking;
-	int16 t = envTotalLevel << 8;
-	if (envAttackRate == 127) {
-		envStep = 0;
-	} else if (envAttackRate) {
-		envStep = t / envAttackRate;
-		envCurrentLevel = 1;
-	} else {
-		envCurrentLevel = t;
-		envDecay();
-	}
-}
-
-void TownsAudio_PcmChannel::envDecay() {
-	envState = kEnvDecaying;
-	int16 t = envTotalLevel - envSustainLevel;
-	if (t < 0 || envDecayRate == 127) {
-		envStep = 0;
-	} else if (envDecayRate) {
-		envStep = (t << 8) / envDecayRate;
-	} else {
-		envCurrentLevel = envSustainLevel << 8;
-		envSustain();
-	}
-}
-
-void TownsAudio_PcmChannel::envSustain() {
-	envState = kEnvSustaining;
-	if (envSustainLevel && envSustainRate)
-		envStep = (envSustainRate == 127) ? 0 : (envCurrentLevel / envSustainRate) >> 1;
-	else
-		envStep = envCurrentLevel = 1;
-}
-
-void TownsAudio_PcmChannel::envRelease() {
-	envState = kEnvReleasing;
-	if (envReleaseRate == 127)
-		envStep = 0;
-	else if (envReleaseRate)
-		envStep = envCurrentLevel / envReleaseRate;
-	else
-		envStep = envCurrentLevel = 1;
-}
-
-TownsAudio_WaveTable::TownsAudio_WaveTable() {
-	data = 0;
-	clear();
-}
-
-TownsAudio_WaveTable::~TownsAudio_WaveTable() {
-	clear();
-}
-
-void TownsAudio_WaveTable::readHeader(const uint8 *buffer) {
-	memcpy(name, buffer, 8);
-	name[8] = 0;
-	id = READ_LE_UINT32(&buffer[8]);
-	size = READ_LE_UINT32(&buffer[12]);
-	loopStart = READ_LE_UINT32(&buffer[16]);
-	loopLen = READ_LE_UINT32(&buffer[20]);
-	rate = READ_LE_UINT16(&buffer[24]);
-	rateOffs = READ_LE_UINT16(&buffer[26]);
-	baseNote = READ_LE_UINT32(&buffer[28]);
-}
-
-void TownsAudio_WaveTable::readData(const uint8 *buffer) {
-	if (!size)
-		return;
-
-	delete[] data;
-	data = new int8[size];
-
-	const int8 *src = (const int8 *)buffer;
-	int8 *dst = data;
-	for (uint32 i = 0; i < size; i++)
-		*dst++ = *src & 0x80 ? (*src++ & 0x7f) : -*src++;
-}
-
-void TownsAudio_WaveTable::clear() {
-	name[0] = name[8] = 0;
-	id = -1;
-	size = 0;
-	loopStart = 0;
-	loopLen = 0;
-	rate = 0;
-	rateOffs = 0;
-	baseNote = 0;
-	delete[] data;
-	data = 0;
-}
-
diff --git a/sound/softsynth/fmtowns_pc98/towns_audio.h b/sound/softsynth/fmtowns_pc98/towns_audio.h
deleted file mode 100644
index 95fb1ded59..0000000000
--- a/sound/softsynth/fmtowns_pc98/towns_audio.h
+++ /dev/null
@@ -1,179 +0,0 @@
-/* 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 TOWNS_AUDIO_H
-#define TOWNS_AUDIO_H
-
-#include "sound/softsynth/fmtowns_pc98/towns_pc98_fmsynth.h"
-
-class TownsAudioInterfacePluginDriver {
-public:
-	virtual ~TownsAudioInterfacePluginDriver() {}
-	virtual void timerCallback(int timerId) = 0;
-};
-
-class TownsAudio_PcmChannel;
-class TownsAudio_WaveTable;
-
-class TownsAudioInterface : public TownsPC98_FmSynth {
-public:
-	TownsAudioInterface(Audio::Mixer *mixer, TownsAudioInterfacePluginDriver *driver);
-	~TownsAudioInterface();
-
-	bool init();
-
-	int callback(int command, ...);
-
-	void setMusicVolume(int volume);
-	void setSoundEffectVolume(int volume);
-	// Defines the channels used as sound effect channels for the purpose of ScummVM GUI volume control.
-	// The first 6 bits are the 6 fm channels. The next 8 bits are pcm channels.
-	void setSoundEffectChanMask(int mask);
-
-private:
-	void nextTickEx(int32 *buffer, uint32 bufferSize);
-
-	void timerCallbackA();
-	void timerCallbackB();
-
-	typedef int (TownsAudioInterface::*TownsAudioIntfCallback)(va_list &);
-	const TownsAudioIntfCallback *_intfOpcodes;
-
-	int intf_reset(va_list &args);
-	int intf_keyOn(va_list &args);
-	int intf_keyOff(va_list &args);
-	int intf_setPanPos(va_list &args);
-	int intf_setInstrument(va_list &args);
-	int intf_loadInstrument(va_list &args);
-	int intf_setPitch(va_list &args);
-	int intf_setLevel(va_list &args);
-	int intf_chanOff(va_list &args);
-	int intf_writeReg(va_list &args);
-	int intf_writeRegBuffer(va_list &args);
-	int intf_readRegBuffer(va_list &args);
-	int intf_setTimerA(va_list &args);
-	int intf_setTimerB(va_list &args);
-	int intf_enableTimerA(va_list &args);
-	int intf_enableTimerB(va_list &args);
-	int intf_loadSamples(va_list &args);
-	int intf_reserveEffectChannels(va_list &args);
-	int intf_loadWaveTable(va_list &args);
-	int intf_unloadWaveTable(va_list &args);
-	int intf_pcmPlayEffect(va_list &args);
-	int intf_pcmChanOff(va_list &args);
-	int intf_pcmEffectPlaying(va_list &args);
-	int intf_fmKeyOn(va_list &args);
-	int intf_fmKeyOff(va_list &args);
-	int intf_fmSetPanPos(va_list &args);
-	int intf_fmSetInstrument(va_list &args);
-	int intf_fmLoadInstrument(va_list &args);
-	int intf_fmSetPitch(va_list &args);
-	int intf_fmSetLevel(va_list &args);
-	int intf_fmReset(va_list &args);
-	int intf_setOutputVolume(va_list &args);
-	int intf_resetOutputVolume(va_list &args);
-	int intf_updateOutputVolume(va_list &args);
-	int intf_cdaToggle(va_list &args);
-	int intf_pcmUpdateEnvelopeGenerator(va_list &args);
-
-	int intf_notImpl(va_list &args);
-
-	void fmReset();
-	int fmKeyOn(int chan, int note, int velo);
-	int fmKeyOff(int chan);
-	int fmChanOff(int chan);
-	int fmSetPanPos(int chan, int mode);
-	int fmSetInstrument(int chan, int instrId);
-	int fmLoadInstrument(int instrId, const uint8 *data);
-	int fmSetPitch(int chan, int pitch);
-	int fmSetLevel(int chan, int lvl);
-
-	void bufferedWriteReg(uint8 part, uint8 regAddress, uint8 value);
-
-	uint8 _fmChanPlaying;
-	uint8 _fmChanNote[6];
-	int16 _fmChanPitch[6];
-
-	uint8 *_fmSaveReg[2];
-	uint8 *_fmInstruments;
-
-	void pcmReset();
-	int pcmKeyOn(int chan, int note, int velo);
-	int pcmKeyOff(int chan);
-	int pcmChanOff(int chan);
-	int pcmSetPanPos(int chan, int mode);
-	int pcmSetInstrument(int chan, int instrId);
-	int pcmLoadInstrument(int instrId, const uint8 *data);
-	int pcmSetPitch(int chan, int pitch);
-	int pcmSetLevel(int chan, int lvl);
-	void pcmUpdateEnvelopeGenerator(int chan);
-
-	TownsAudio_PcmChannel *_pcmChan;
-	uint8 _pcmChanOut;
-	uint8 _pcmChanReserved;
-	uint8 _pcmChanKeyPressed;
-	uint8 _pcmChanEffectPlaying;
-	uint8 _pcmChanKeyPlaying;
-
-	uint8 _pcmChanNote[8];
-	uint8 _pcmChanVelo[8];
-	uint8 _pcmChanLevel[8];
-
-	uint8 _numReservedChannels;
-	uint8 *_pcmInstruments;
-
-	TownsAudio_WaveTable *_waveTables;
-	uint8 _numWaveTables;
-	uint32 _waveTablesTotalDataSize;
-
-	void pcmCalcPhaseStep(TownsAudio_PcmChannel *p, TownsAudio_WaveTable *w);
-
-	void updateOutputVolume();
-	uint8 _outputVolumeFlags;
-	uint8 _outputLevel[16];
-	uint8 _outputMuteFlags;
-
-	const float _baserate;
-	uint32 _timerBase;
-	uint32 _tickLength;
-	uint32 _timer;
-
-	uint16 _musicVolume;
-	uint16 _sfxVolume;
-	int _pcmSfxChanMask;
-
-	TownsAudioInterfacePluginDriver *_drv;
-	bool _ready;
-
-	static const uint8 _chanFlags[];
-	static const uint16 _frequency[];
-	static const uint8 _carrier[];
-	static const uint8 _fmDefaultInstrument[];
-	static const uint16 _pcmPhase1[];
-	static const uint16 _pcmPhase2[];
-};
-
-#endif
-
diff --git a/sound/softsynth/fmtowns_pc98/towns_euphony.cpp b/sound/softsynth/fmtowns_pc98/towns_euphony.cpp
deleted file mode 100644
index 3a691c407d..0000000000
--- a/sound/softsynth/fmtowns_pc98/towns_euphony.cpp
+++ /dev/null
@@ -1,905 +0,0 @@
-/* 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 "sound/softsynth/fmtowns_pc98/towns_euphony.h"
-#include "common/endian.h"
-
-TownsEuphonyDriver::TownsEuphonyDriver(Audio::Mixer *mixer) : _activeChannels(0), _sustainChannels(0),
-	_assignedChannels(0), _paraCount(0), _command(0), _tEnable(0), _tMode(0), _tOrdr(0), _tLevel(0),
-	_tTranspose(0), _musicPos(0), _musicStart(0), _playing(false), _eventBuffer(0), _bufferedEventsCount(0),
-	_tempoControlMode(0) {
-	_para[0] = _para[1] = 0;
-	_intf = new TownsAudioInterface(mixer, this);
-	resetTempo();
-}
-
-TownsEuphonyDriver::~TownsEuphonyDriver() {
-	delete _intf;
-	delete[] _activeChannels;
-	delete[] _sustainChannels;
-	delete[] _assignedChannels;
-	delete[] _tEnable;
-	delete[] _tMode;
-	delete[] _tOrdr;
-	delete[] _tLevel;
-	delete[] _tTranspose;
-}
-
-bool TownsEuphonyDriver::init() {
-	if (!_intf->init())
-		return false;
-
-	_activeChannels = new int8[16];
-	_sustainChannels = new int8[16];
-	_assignedChannels = new ActiveChannel[128];
-	_eventBuffer = new DlEvent[64];
-
-	_tEnable = new uint8[32];
-	_tMode = new uint8[32];
-	_tOrdr = new uint8[32];
-	_tLevel = new int8[32];
-	_tTranspose = new int8[32];
-
-	reset();
-
-	return true;
-}
-
-void TownsEuphonyDriver::reset() {
-	_intf->callback(0);
-
-	_intf->callback(74);
-	_intf->callback(70);
-	_intf->callback(75, 3);
-
-	setTimerA(true, 1);
-	setTimerA(false, 1);
-	setTimerB(true, 221);
-
-	_paraCount = _command = _para[0] = _para[1] = 0;
-	memset(_sustainChannels, 0, 16);
-	memset(_activeChannels, -1, 16);
-	for (int i = 0; i < 128; i++) {
-		_assignedChannels[i].chan = _assignedChannels[i].next = -1;
-		_assignedChannels[i].note = _assignedChannels[i].sub = 0;
-	}
-
-	int e = 0;
-	for (int i = 0; i < 6; i++)
-		assignChannel(i, e++);
-	for (int i = 0x40; i < 0x48; i++)
-		assignChannel(i, e++);
-
-	resetTables();
-
-	memset(_eventBuffer, 0, 64 * sizeof(DlEvent));
-	_bufferedEventsCount = 0;
-
-	_playing = _endOfTrack = _suspendParsing = _loop = false;
-	_elapsedEvents = 0;
-	_tempoDiff = 0;
-
-	resetTempo();
-
-	if (_tempoControlMode == 1) {
-		//if (///)
-		//  return;
-		setTempoIntern(_defaultTempo);
-	} else {
-		setTempoIntern(_defaultTempo);
-	}
-
-	resetControl();
-}
-
-void TownsEuphonyDriver::loadInstrument(int chanType, int id, const uint8 *data) {
-	_intf->callback(5, chanType, id, data);
-}
-
-void TownsEuphonyDriver::loadWaveTable(const uint8 *data) {
-	_intf->callback(34, data);
-}
-
-void TownsEuphonyDriver::unloadWaveTable(int id) {
-	_intf->callback(35, id);
-}
-
-void TownsEuphonyDriver::reserveSoundEffectChannels(int num) {
-	_intf->callback(33, num);
-	uint32 volMask = 0;
-
-	if (num > 8)
-		return;
-
-	for (uint32 v = 1 << 13; num; num--) {
-		volMask |= v;
-		v >>= 1;
-	}
-
-	_intf->setSoundEffectChanMask(volMask);
-}
-
-int TownsEuphonyDriver::setMusicTempo(int tempo) {
-	if (tempo > 250)
-		return 3;
-	_defaultTempo = tempo;
-	_trackTempo = tempo;
-	setTempoIntern(tempo);
-	return 0;
-}
-
-int TownsEuphonyDriver::startMusicTrack(const uint8 *data, int trackSize, int startTick) {
-	if (_playing)
-		return 2;
-
-	_musicPos = _musicStart = data;
-	_defaultBaseTickLen = _baseTickLen = startTick;
-	_musicTrackSize = trackSize;
-	_timeStampBase = _timeStampDest = 0;
-	_tickCounter = 0;
-	_playing = true;
-
-	return 0;
-}
-
-void TownsEuphonyDriver::setMusicLoop(bool loop) {
-	_loop = loop;
-}
-
-void TownsEuphonyDriver::stopParser() {
-	if (_playing) {
-		_playing = false;
-		_pulseCount = 0;
-		_endOfTrack = false;
-		flushEventBuffer();
-		resetControl();
-	}
-}
-
-void TownsEuphonyDriver::continueParsing() {
-	_suspendParsing = false;
-}
-
-void TownsEuphonyDriver::playSoundEffect(int chan, int note, int velo, const uint8 *data) {
-	_intf->callback(37, chan, note, velo, data);
-}
-
-void TownsEuphonyDriver::stopSoundEffect(int chan) {
-	_intf->callback(39, chan);
-}
-
-bool TownsEuphonyDriver::soundEffectIsPlaying(int chan) {
-	return _intf->callback(40, chan) ? true : false;
-}
-
-void TownsEuphonyDriver::chanPanPos(int chan, int mode) {
-	_intf->callback(3, chan, mode);
-}
-
-void TownsEuphonyDriver::chanPitch(int chan, int pitch) {
-	_intf->callback(7, chan, pitch);
-}
-
-void TownsEuphonyDriver::chanVolume(int chan, int vol) {
-	_intf->callback(8, chan, vol);
-}
-
-void TownsEuphonyDriver::setOutputVolume(int mode, int volLeft, int volRight) {
-	_intf->callback(67, mode, volLeft, volRight);
-}
-
-int TownsEuphonyDriver::chanEnable(int tableEntry, int val) {
-	if (tableEntry > 31)
-		return 3;
-	_tEnable[tableEntry] = val;
-	return 0;
-}
-
-int TownsEuphonyDriver::chanMode(int tableEntry, int val) {
-	if (tableEntry > 31)
-		return 3;
-	_tMode[tableEntry] = val;
-	return 0;
-}
-
-int TownsEuphonyDriver::chanOrdr(int tableEntry, int val) {
-	if (tableEntry > 31)
-		return 3;
-	if (val < 16)
-		_tOrdr[tableEntry] = val;
-	return 0;
-}
-
-int TownsEuphonyDriver::chanVolumeShift(int tableEntry, int val) {
-	if (tableEntry > 31)
-		return 3;
-	if (val <= 40)
-		_tLevel[tableEntry] = (int8)(val & 0xff);
-	return 0;
-}
-
-int TownsEuphonyDriver::chanNoteShift(int tableEntry, int val) {
-	if (tableEntry > 31)
-		return 3;
-	if (val <= 40)
-		_tTranspose[tableEntry] = (int8)(val & 0xff);
-	return 0;
-}
-
-int TownsEuphonyDriver::assignChannel(int chan, int tableEntry) {
-	if (tableEntry > 15 || chan > 127 || chan < 0)
-		return 3;
-
-	ActiveChannel *a = &_assignedChannels[chan];
-	if (a->chan == tableEntry)
-		return 0;
-
-	if (a->chan != -1) {
-		int8 *b = &_activeChannels[a->chan];
-		while (*b != chan) {
-			b = &_assignedChannels[*b].next;
-			if (*b == -1 && *b != chan)
-				return 3;
-		}
-
-		*b = a->next;
-
-		if (a->note)
-			_intf->callback(2, chan);
-
-		a->chan = a->next = -1;
-		a->note = 0;
-	}
-
-	a->next = _activeChannels[tableEntry];
-	_activeChannels[tableEntry] = chan;
-	a->chan = tableEntry;
-	a->note = a->sub = 0;
-
-	return 0;
-}
-
-void TownsEuphonyDriver::timerCallback(int timerId) {
-	switch (timerId) {
-	case 0:
-		updatePulseCount();
-		while (_pulseCount > 0) {
-			--_pulseCount;
-			updateTimeStampBase();
-			if (!_playing)
-				continue;
-			updateEventBuffer();
-			updateParser();
-			updateCheckEot();
-		}
-		break;
-	default:
-		break;
-	}
-}
-
-void TownsEuphonyDriver::setMusicVolume(int volume) {
-	_intf->setMusicVolume(volume);
-}
-
-void TownsEuphonyDriver::setSoundEffectVolume(int volume) {
-	_intf->setSoundEffectVolume(volume);
-}
-
-void TownsEuphonyDriver::resetTables() {
-	memset(_tEnable, 0xff, 32);
-	memset(_tMode, 0xff, 16);
-	memset(_tMode + 16, 0, 16);
-	for (int i = 0; i < 32; i++)
-		_tOrdr[i] = i & 0x0f;
-	memset(_tLevel, 0, 32);
-	memset(_tTranspose, 0, 32);
-}
-
-void TownsEuphonyDriver::resetTempo() {
-	_defaultBaseTickLen = _baseTickLen = 0x33;
-	_pulseCount = 0;
-	_extraTimingControlRemainder = 0;
-	_extraTimingControl = 16;
-	_tempoModifier = 0;
-	_timeStampDest = 0;
-	_deltaTicks = 0;
-	_tickCounter = 0;
-	_defaultTempo = 90;
-	_trackTempo = 90;
-}
-
-void TownsEuphonyDriver::setTempoIntern(int tempo) {
-	tempo = CLIP(tempo + _tempoModifier, 0, 500);
-	if (_tempoControlMode == 0) {
-		_timerSetting = 34750 / (tempo + 30);
-		_extraTimingControl = 16;
-
-		while (_timerSetting < 126) {
-			_timerSetting <<= 1;
-			_extraTimingControl <<= 1;
-		}
-
-		while (_timerSetting > 383) {
-			_timerSetting >>= 1;
-			_extraTimingControl >>= 1;
-		}
-
-		setTimerA(true, -(_timerSetting - 2));
-
-	} else if (_tempoControlMode == 1) {
-		_timerSetting = 312500 / (tempo + 30);
-		_extraTimingControl = 16;
-		while (_timerSetting < 1105) {
-			_timerSetting <<= 1;
-			_extraTimingControl <<= 1;
-		}
-
-	} else if (_tempoControlMode == 2) {
-		_timerSetting = 625000 / (tempo + 30);
-		_extraTimingControlRemainder = 0;
-	}
-}
-
-void TownsEuphonyDriver::setTimerA(bool enable, int tempo) {
-	_intf->callback(21, enable ? 255 : 0, tempo);
-}
-
-void TownsEuphonyDriver::setTimerB(bool enable, int tempo) {
-	_intf->callback(22, enable ? 255 : 0, tempo);
-}
-
-void TownsEuphonyDriver::updatePulseCount() {
-	int tc = _extraTimingControl + _extraTimingControlRemainder;
-	_extraTimingControlRemainder = tc & 0x0f;
-	tc >>= 4;
-	_tempoDiff -= tc;
-
-	while (_tempoDiff < 0) {
-		_elapsedEvents++;
-		_tempoDiff += 4;
-	}
-
-	if (_playing && !_suspendParsing)
-		_pulseCount += tc;
-}
-
-void TownsEuphonyDriver::updateTimeStampBase() {
-	static const uint16 table[] = { 0x180, 0xC0, 0x80, 0x60, 0x40, 0x30, 0x20, 0x18 };
-	if ((uint32)(table[_baseTickLen >> 4] * ((_baseTickLen & 0x0f) + 1)) > ++_tickCounter)
-		return;
-	++_timeStampDest;
-	_tickCounter = 0;
-	_deltaTicks = 0;
-}
-
-void TownsEuphonyDriver::updateParser() {
-	for (bool loop = true; loop;) {
-		uint8 cmd = _musicPos[0];
-
-		if (cmd == 0xff || cmd == 0xf7) {
-			jumpNextLoop();
-
-		} else if (cmd < 0x90) {
-			_endOfTrack = true;
-			flushEventBuffer();
-			loop = false;
-
-		} else if (_timeStampBase > _timeStampDest) {
-			loop = false;
-
-		} else {
-			if (_timeStampBase == _timeStampDest) {
-				uint16 timeStamp = READ_LE_UINT16(&_musicPos[2]);
-				uint8 l = (timeStamp & 0xff) + (timeStamp & 0xff);
-				timeStamp = ((timeStamp & 0xff00) | l) >> 1;
-				if (timeStamp > _tickCounter)
-					loop = false;
-			}
-
-			if (loop) {
-				if (parseNext())
-					loop = false;
-			}
-		}
-	}
-}
-
-void TownsEuphonyDriver::updateCheckEot() {
-	if (!_endOfTrack || _bufferedEventsCount)
-		return;
-	stopParser();
-}
-
-bool TownsEuphonyDriver::parseNext() {
-#define OPC(x) &TownsEuphonyDriver::evt##x
-	static const EuphonyOpcode opcodes[] = {
-		OPC(NotImpl),
-		OPC(SetupNote),
-		OPC(PolyphonicAftertouch),
-		OPC(ControlPitch),
-		OPC(InstrumentChanAftertouch),
-		OPC(InstrumentChanAftertouch),
-		OPC(ControlPitch)
-	};
-#undef OPC
-
-	uint cmd = _musicPos[0];
-	if (cmd != 0xfe && cmd != 0xfd) {
-		if (cmd >= 0xf0) {
-			cmd &= 0x0f;
-			if (cmd == 0)
-				evtLoadInstrument();
-			else if (cmd == 2)
-				evtAdvanceTimestampOffset();
-			else if (cmd == 8)
-				evtTempo();
-			else if (cmd == 12)
-				evtModeOrdrChange();
-			jumpNextLoop();
-			return false;
-
-		} else if (!(this->*opcodes[(cmd - 0x80) >> 4])()) {
-			jumpNextLoop();
-			return false;
-		}
-	}
-
-	if (cmd == 0xfd) {
-		_suspendParsing = true;
-		return true;
-	}
-
-	if (!_loop) {
-		_endOfTrack = true;
-		return true;
-	}
-
-	_endOfTrack = false;
-	_musicPos = _musicStart;
-	_timeStampBase = _timeStampDest = _tickCounter = 0;
-	_baseTickLen = _defaultBaseTickLen;
-
-	return false;
-}
-
-void TownsEuphonyDriver::jumpNextLoop() {
-	_musicPos += 6;
-	if (_musicPos >= _musicStart + _musicTrackSize)
-		_musicPos = _musicStart;
-}
-
-void TownsEuphonyDriver::updateEventBuffer() {
-	DlEvent *e = _eventBuffer;
-	for (int i = _bufferedEventsCount; i; e++) {
-		if (e->evt == 0)
-			continue;
-		if (--e->len) {
-			--i;
-			continue;
-		}
-		processBufferNote(e->mode, e->evt, e->note, e->velo);
-		e->evt = 0;
-		--i;
-		--_bufferedEventsCount;
-	}
-}
-
-void TownsEuphonyDriver::flushEventBuffer() {
-	DlEvent *e = _eventBuffer;
-	for (int i = _bufferedEventsCount; i; e++) {
-		if (e->evt == 0)
-			continue;
-		processBufferNote(e->mode, e->evt, e->note, e->velo);
-		e->evt = 0;
-		--i;
-		--_bufferedEventsCount;
-	}
-}
-
-void TownsEuphonyDriver::processBufferNote(int mode, int evt, int note, int velo) {
-	if (!velo)
-		evt &= 0x8f;
-	sendEvent(mode, evt);
-	sendEvent(mode, note);
-	sendEvent(mode, velo);
-}
-
-void TownsEuphonyDriver::resetControl() {
-	for (int i = 0; i < 32; i++) {
-		if (_tOrdr[i] > 15) {
-			for (int ii = 0; ii < 16; ii++)
-				resetControlIntern(_tMode[i], ii);
-		} else {
-			resetControlIntern(_tMode[i], _tOrdr[i]);
-		}
-	}
-}
-
-void TownsEuphonyDriver::resetControlIntern(int mode, int chan) {
-	sendEvent(mode, 0xb0 | chan);
-	sendEvent(mode, 0x40);
-	sendEvent(mode, 0);
-	sendEvent(mode, 0xb0 | chan);
-	sendEvent(mode, 0x7b);
-	sendEvent(mode, 0);
-	sendEvent(mode, 0xb0 | chan);
-	sendEvent(mode, 0x79);
-	sendEvent(mode, 0x40);
-}
-
-uint8 TownsEuphonyDriver::appendEvent(uint8 evt, uint8 chan) {
-	if (evt >= 0x80 && evt < 0xf0 && _tOrdr[chan] < 16)
-		return (evt & 0xf0) | _tOrdr[chan];
-	return evt;
-}
-
-void TownsEuphonyDriver::sendEvent(uint8 mode, uint8 command) {
-	if (mode == 0) {
-		// warning("TownsEuphonyDriver: Mode 0 not implemented");
-
-	} else if (mode == 0x10) {
-		warning("TownsEuphonyDriver: Mode 0x10 not implemented");
-
-	} else if (mode == 0xff) {
-		if (command >= 0xf0) {
-			_paraCount = 1;
-			_command = 0;
-		} else if (command >= 0x80) {
-			_paraCount = 1;
-			_command = command;
-		} else if (_command >= 0x80) {
-			switch ((_command - 0x80) >> 4) {
-			case 0:
-				if (_paraCount < 2) {
-					_paraCount++;
-					_para[0] = command;
-				} else {
-					_paraCount = 1;
-					_para[1] = command;
-					sendNoteOff();
-				}
-				break;
-
-			case 1:
-				if (_paraCount < 2) {
-					_paraCount++;
-					_para[0] = command;
-				} else {
-					_paraCount = 1;
-					_para[1] = command;
-					if (command)
-						sendNoteOn();
-					else
-						sendNoteOff();
-				}
-				break;
-
-			case 2:
-				if (_paraCount < 2) {
-					_paraCount++;
-					_para[0] = command;
-				} else {
-					_paraCount = 1;
-				}
-				break;
-
-			case 3:
-				if (_paraCount < 2) {
-					_paraCount++;
-					_para[0] = command;
-				} else {
-					_paraCount = 1;
-					_para[1] = command;
-
-					if (_para[0] == 7)
-						sendChanVolume();
-					else if (_para[0] == 10)
-						sendPanPosition();
-					else if (_para[0] == 64)
-						sendAllNotesOff();
-				}
-				break;
-
-			case 4:
-				_paraCount = 1;
-				_para[0] = command;
-				sendSetInstrument();
-				break;
-
-			case 5:
-				_paraCount = 1;
-				_para[0] = command;
-				break;
-
-			case 6:
-				if (_paraCount < 2) {
-					_paraCount++;
-					_para[0] = command;
-				} else {
-					_paraCount = 1;
-					_para[1] = command;
-					sendPitch();
-				}
-				break;
-			}
-		}
-	}
-}
-
-bool TownsEuphonyDriver::evtSetupNote() {
-	if (_musicPos[1] > 31)
-		return false;
-	if (!_tEnable[_musicPos[1]]) {
-		jumpNextLoop();
-		return (_musicPos[0] == 0xfe || _musicPos[0] == 0xfd) ? true : false;
-	}
-	uint8 evt = appendEvent(_musicPos[0], _musicPos[1]);
-	uint8 mode = _tMode[_musicPos[1]];
-	uint8 note = _musicPos[4];
-	uint8 velo = _musicPos[5];
-
-	sendEvent(mode, evt);
-	sendEvent(mode, applyNoteShift(note));
-	sendEvent(mode, applyVolumeShift(velo));
-
-	jumpNextLoop();
-	if (_musicPos[0] == 0xfe || _musicPos[0] == 0xfd)
-		return true;
-
-	velo = _musicPos[5];
-	uint16 len = ((((_musicPos[1] << 4) | (_musicPos[2] << 8)) >> 4) & 0xff) | ((((_musicPos[3] << 4) | (_musicPos[4] << 8)) >> 4) << 8);
-
-	int i = 0;
-	for (; i < 64; i++) {
-		if (_eventBuffer[i].evt == 0)
-			break;
-	}
-
-	if (i == 64) {
-		processBufferNote(mode, evt, note, velo);
-	} else {
-		_eventBuffer[i].evt = evt;
-		_eventBuffer[i].mode = mode;
-		_eventBuffer[i].note = note;
-		_eventBuffer[i].velo = velo;
-		_eventBuffer[i].len = len ? len : 1;
-		_bufferedEventsCount++;
-	}
-
-	return false;
-}
-
-bool TownsEuphonyDriver::evtPolyphonicAftertouch() {
-	if (_musicPos[1] > 31)
-		return false;
-	if (!_tEnable[_musicPos[1]])
-		return false;
-
-	uint8 evt = appendEvent(_musicPos[0], _musicPos[1]);
-	uint8 mode = _tMode[_musicPos[1]];
-
-	sendEvent(mode, evt);
-	sendEvent(mode, applyNoteShift(_musicPos[4]));
-	sendEvent(mode, _musicPos[5]);
-
-	return false;
-}
-
-bool TownsEuphonyDriver::evtControlPitch() {
-	if (_musicPos[1] > 31)
-		return false;
-	if (!_tEnable[_musicPos[1]])
-		return false;
-
-	uint8 evt = appendEvent(_musicPos[0], _musicPos[1]);
-	uint8 mode = _tMode[_musicPos[1]];
-
-	sendEvent(mode, evt);
-	sendEvent(mode, _musicPos[4]);
-	sendEvent(mode, _musicPos[5]);
-
-	return false;
-}
-
-bool TownsEuphonyDriver::evtInstrumentChanAftertouch() {
-	if (_musicPos[1] > 31)
-		return false;
-	if (!_tEnable[_musicPos[1]])
-		return false;
-
-	uint8 evt = appendEvent(_musicPos[0], _musicPos[1]);
-	uint8 mode = _tMode[_musicPos[1]];
-
-	sendEvent(mode, evt);
-	sendEvent(mode, _musicPos[4]);
-
-	return false;
-}
-
-bool TownsEuphonyDriver::evtLoadInstrument() {
-	return false;
-}
-
-bool TownsEuphonyDriver::evtAdvanceTimestampOffset() {
-	++_timeStampBase;
-	_baseTickLen = _musicPos[1];
-	return false;
-}
-
-bool TownsEuphonyDriver::evtTempo() {
-	uint8 l = _musicPos[4] << 1;
-	_trackTempo = (l | (_musicPos[5] << 8)) >> 1;
-	setTempoIntern(_trackTempo);
-	return false;
-}
-
-bool TownsEuphonyDriver::evtModeOrdrChange() {
-	if (_musicPos[1] > 31)
-		return false;
-	if (!_tEnable[_musicPos[1]])
-		return false;
-
-	if (_musicPos[4] == 1)
-		_tMode[_musicPos[1]] = _musicPos[5];
-	else if (_musicPos[4] == 2)
-		_tOrdr[_musicPos[1]] = _musicPos[5];
-
-	return false;
-}
-
-uint8 TownsEuphonyDriver::applyNoteShift(uint8 in) {
-	int out = _tTranspose[_musicPos[1]];
-	if (!out)
-		return in;
-	out += (in & 0x7f);
-
-	if (out > 127)
-		out -= 12;
-
-	if (out < 0)
-		out += 12;
-
-	return out & 0xff;
-}
-
-uint8 TownsEuphonyDriver::applyVolumeShift(uint8 in) {
-	int out = _tLevel[_musicPos[1]];
-	out += (in & 0x7f);
-	out = CLIP(out, 1, 127);
-
-	return out & 0xff;
-}
-
-void TownsEuphonyDriver::sendNoteOff() {
-	int8 *chan = &_activeChannels[_command & 0x0f];
-	if (*chan == -1)
-		return;
-
-	while (_assignedChannels[*chan].note != _para[0]) {
-		chan = &_assignedChannels[*chan].next;
-		if (*chan == -1)
-			return;
-	}
-
-	if (_sustainChannels[_command & 0x0f]) {
-		_assignedChannels[*chan].note |= 0x80;
-	} else {
-		_assignedChannels[*chan].note = 0;
-		_intf->callback(2, *chan);
-	}
-}
-
-void TownsEuphonyDriver::sendNoteOn() {
-	if (!_para[0])
-		return;
-	int8 *chan = &_activeChannels[_command & 0x0f];
-	if (*chan == -1)
-		return;
-
-	do {
-		_assignedChannels[*chan].sub++;
-		chan = &_assignedChannels[*chan].next;
-	} while (*chan != -1);
-
-	chan = &_activeChannels[_command & 0x0f];
-
-	int d = 0;
-	int c = 0;
-	bool found = false;
-
-	do {
-		if (!_assignedChannels[*chan].note) {
-			found = true;
-			break;
-		}
-		if (d <= _assignedChannels[*chan].sub) {
-			c = *chan;
-			d = _assignedChannels[*chan].sub;
-		}
-		chan = &_assignedChannels[*chan].next;
-	} while (*chan != -1);
-
-	if (found)
-		c = *chan;
-	else
-		_intf->callback(2, c);
-
-	_assignedChannels[c].note = _para[0];
-	_assignedChannels[c].sub = 0;
-	_intf->callback(1, c, _para[0], _para[1]);
-}
-
-void TownsEuphonyDriver::sendChanVolume() {
-	int8 *chan = &_activeChannels[_command & 0x0f];
-	while (*chan != -1) {
-		_intf->callback(8, *chan, _para[1] & 0x7f);
-		chan = &_assignedChannels[*chan].next;
-	}
-}
-
-void TownsEuphonyDriver::sendPanPosition() {
-	int8 *chan = &_activeChannels[_command & 0x0f];
-	while (*chan != -1) {
-		_intf->callback(3, *chan, _para[1] & 0x7f);
-		chan = &_assignedChannels[*chan].next;
-	}
-}
-
-void TownsEuphonyDriver::sendAllNotesOff() {
-	if (_para[1] > 63) {
-		_sustainChannels[_command & 0x0f] = -1;
-		return;
-	}
-
-	_sustainChannels[_command & 0x0f] = 0;
-	int8 *chan = &_activeChannels[_command & 0x0f];
-	while (*chan != -1) {
-		if (_assignedChannels[*chan].note & 0x80) {
-			_assignedChannels[*chan].note = 0;
-			_intf->callback(2, *chan);
-		}
-		chan = &_assignedChannels[*chan].next;
-	}
-}
-
-void TownsEuphonyDriver::sendSetInstrument() {
-	int8 *chan = &_activeChannels[_command & 0x0f];
-	while (*chan != -1) {
-		_intf->callback(4, *chan, _para[0]);
-		_intf->callback(7, *chan, 0);
-		chan = &_assignedChannels[*chan].next;
-	}
-}
-
-void TownsEuphonyDriver::sendPitch() {
-	int8 *chan = &_activeChannels[_command & 0x0f];
-	while (*chan != -1) {
-		_para[0] += _para[0];
-		int16 pitch = (((READ_LE_UINT16(_para)) >> 1) & 0x3fff) - 0x2000;
-		_intf->callback(7, *chan, pitch);
-		chan = &_assignedChannels[*chan].next;
-	}
-}
diff --git a/sound/softsynth/fmtowns_pc98/towns_euphony.h b/sound/softsynth/fmtowns_pc98/towns_euphony.h
deleted file mode 100644
index fa1f8ba496..0000000000
--- a/sound/softsynth/fmtowns_pc98/towns_euphony.h
+++ /dev/null
@@ -1,187 +0,0 @@
-/* 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 TOWNS_EUP_H
-#define TOWNS_EUP_H
-
-#include "sound/softsynth/fmtowns_pc98/towns_audio.h"
-
-class TownsEuphonyDriver : public TownsAudioInterfacePluginDriver {
-public:
-	TownsEuphonyDriver(Audio::Mixer *mixer);
-	virtual ~TownsEuphonyDriver();
-
-	bool init();
-	void reset();
-
-	void loadInstrument(int chanType, int id, const uint8 *data);
-	void loadWaveTable(const uint8 *data);
-	void unloadWaveTable(int id);
-	void reserveSoundEffectChannels(int num);
-
-	int setMusicTempo(int tempo);
-	int startMusicTrack(const uint8 *data, int trackSize, int startTick);
-	void setMusicLoop(bool loop);
-	void stopParser();
-	bool parserIsPlaying() {return _playing; }
-	void continueParsing();
-
-	void playSoundEffect(int chan, int note, int velo, const uint8 *data);
-	void stopSoundEffect(int chan);
-	bool soundEffectIsPlaying(int chan);
-
-	void chanPanPos(int chan, int mode);
-	void chanPitch(int chan, int pitch);
-	void chanVolume(int chan, int vol);
-
-	void setOutputVolume(int chanType, int volLeft, int volRight);
-
-	int chanEnable(int tableEntry, int val);
-	int chanMode(int tableEntry, int val);
-	int chanOrdr(int tableEntry, int val);
-	int chanVolumeShift(int tableEntry, int val);
-	int chanNoteShift(int tableEntry, int val);
-
-	int assignChannel(int chan, int tableEntry);
-
-	void timerCallback(int timerId);
-
-	void setMusicVolume(int volume);
-	void setSoundEffectVolume(int volume);
-
-	TownsAudioInterface *intf() {
-		return _intf;
-	}
-
-private:
-	void resetTables();
-
-	void resetTempo();
-	void setTempoIntern(int tempo);
-	void setTimerA(bool enable, int tempo);
-	void setTimerB(bool enable, int tempo);
-
-	void updatePulseCount();
-	void updateTimeStampBase();
-	void updateParser();
-	void updateCheckEot();
-
-	bool parseNext();
-	void jumpNextLoop();
-
-	void updateEventBuffer();
-	void flushEventBuffer();
-	void processBufferNote(int mode, int evt, int note, int velo);
-
-	void resetControl();
-	void resetControlIntern(int mode, int chan);
-	uint8 appendEvent(uint8 evt, uint8 chan);
-
-	void sendEvent(uint8 mode, uint8 command);
-
-	typedef bool(TownsEuphonyDriver::*EuphonyOpcode)();
-	bool evtSetupNote();
-	bool evtPolyphonicAftertouch();
-	bool evtControlPitch();
-	bool evtInstrumentChanAftertouch();
-	bool evtLoadInstrument();
-	bool evtAdvanceTimestampOffset();
-	bool evtTempo();
-	bool evtModeOrdrChange();
-	bool evtNotImpl() {
-		return false;
-	}
-
-	uint8 applyNoteShift(uint8 in);
-	uint8 applyVolumeShift(uint8 in);
-
-	void sendNoteOff();
-	void sendNoteOn();
-	void sendChanVolume();
-	void sendPanPosition();
-	void sendAllNotesOff();
-	void sendSetInstrument();
-	void sendPitch();
-
-	int8 *_activeChannels;
-	int8 *_sustainChannels;
-
-	struct ActiveChannel {
-		int8 chan;
-		int8 next;
-		uint8 note;
-		uint8 sub;
-	} *_assignedChannels;
-
-	uint8 *_tEnable;
-	uint8 *_tMode;
-	uint8 *_tOrdr;
-	int8 *_tLevel;
-	int8 *_tTranspose;
-
-	struct DlEvent {
-		uint8 evt;
-		uint8 mode;
-		uint8 note;
-		uint8 velo;
-		uint16 len;
-	} *_eventBuffer;
-	int _bufferedEventsCount;
-
-	uint8 _para[2];
-	uint8 _paraCount;
-	uint8 _command;
-
-	uint8 _defaultBaseTickLen;
-	uint8 _baseTickLen;
-	uint32 _pulseCount;
-	int _tempoControlMode;
-	int _extraTimingControlRemainder;
-	int _extraTimingControl;
-	int _timerSetting;
-	int8 _tempoDiff;
-	int _tempoModifier;
-	uint32 _timeStampDest;
-	uint32 _timeStampBase;
-	int8 _elapsedEvents;
-	uint8 _deltaTicks;
-	uint32 _tickCounter;
-	uint8 _defaultTempo;
-	int _trackTempo;
-
-	bool _loop;
-	bool _playing;
-	bool _endOfTrack;
-	bool _suspendParsing;
-
-	const uint8 *_musicStart;
-	const uint8 *_musicPos;
-	uint32 _musicTrackSize;
-
-	TownsAudioInterface *_intf;
-};
-
-#endif
-
diff --git a/sound/softsynth/fmtowns_pc98/towns_pc98_driver.cpp b/sound/softsynth/fmtowns_pc98/towns_pc98_driver.cpp
deleted file mode 100644
index 303f08e6b1..0000000000
--- a/sound/softsynth/fmtowns_pc98/towns_pc98_driver.cpp
+++ /dev/null
@@ -1,1428 +0,0 @@
-/* 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 "sound/softsynth/fmtowns_pc98/towns_pc98_driver.h"
-#include "common/endian.h"
-
-class TownsPC98_MusicChannel {
-public:
-	TownsPC98_MusicChannel(TownsPC98_AudioDriver *driver, uint8 regOffs, uint8 flgs, uint8 num,
-	                       uint8 key, uint8 prt, uint8 id);
-	virtual ~TownsPC98_MusicChannel();
-	virtual void init();
-
-	typedef enum channelState {
-		CHS_RECALCFREQ = 0x01,
-		CHS_KEYOFF     = 0x02,
-		CHS_SSGOFF     = 0x04,
-		CHS_VBROFF     = 0x08,
-		CHS_ALLOFF     = 0x0f,
-		CHS_PROTECT    = 0x40,
-		CHS_EOT        = 0x80
-	} ChannelState;
-
-	virtual void loadData(uint8 *data);
-	virtual void processEvents();
-	virtual void processFrequency();
-	virtual bool processControlEvent(uint8 cmd);
-
-	virtual void keyOn();
-	void keyOff();
-
-	void setOutputLevel();
-	virtual void fadeStep();
-	virtual void reset();
-
-	const uint8 _idFlag;
-
-protected:
-	void setupVibrato();
-	bool processVibrato();
-
-	bool control_dummy(uint8 para);
-	bool control_f0_setPatch(uint8 para);
-	bool control_f1_presetOutputLevel(uint8 para);
-	bool control_f2_setKeyOffTime(uint8 para);
-	bool control_f3_setFreqLSB(uint8 para);
-	bool control_f4_setOutputLevel(uint8 para);
-	bool control_f5_setTempo(uint8 para);
-	bool control_f6_repeatSection(uint8 para);
-	bool control_f7_setupVibrato(uint8 para);
-	bool control_f8_toggleVibrato(uint8 para);
-	bool control_fa_writeReg(uint8 para);
-	virtual bool control_fb_incOutLevel(uint8 para);
-	virtual bool control_fc_decOutLevel(uint8 para);
-	bool control_fd_jump(uint8 para);
-	virtual bool control_ff_endOfTrack(uint8 para);
-
-	uint8 _ticksLeft;
-	uint8 _algorithm;
-	uint8 _instr;
-	uint8 _totalLevel;
-	uint8 _frqBlockMSB;
-	int8 _frqLSB;
-	uint8 _keyOffTime;
-	bool _hold;
-	uint8 *_dataPtr;
-	uint8 _vbrInitDelayHi;
-	uint8 _vbrInitDelayLo;
-	int16 _vbrModInitVal;
-	uint8 _vbrDuration;
-	uint8 _vbrCurDelay;
-	int16 _vbrModCurVal;
-	uint8 _vbrDurLeft;
-	uint16 _frequency;
-	uint8 _block;
-	uint8 _regOffset;
-	uint8 _flags;
-	uint8 _ssgTl;
-	uint8 _ssgStep;
-	uint8 _ssgTicksLeft;
-	uint8 _ssgTargetLvl;
-	uint8 _ssgStartLvl;
-
-	const uint8 _chanNum;
-	const uint8 _keyNum;
-	const uint8 _part;
-
-	TownsPC98_AudioDriver *_drv;
-
-	typedef bool (TownsPC98_MusicChannel::*ControlEventFunc)(uint8 para);
-	const ControlEventFunc *controlEvents;
-};
-
-class TownsPC98_MusicChannelSSG : public TownsPC98_MusicChannel {
-public:
-	TownsPC98_MusicChannelSSG(TownsPC98_AudioDriver *driver, uint8 regOffs,
-	                          uint8 flgs, uint8 num, uint8 key, uint8 prt, uint8 id);
-	virtual ~TownsPC98_MusicChannelSSG() {}
-	void init();
-
-	virtual void loadData(uint8 *data);
-	void processEvents();
-	void processFrequency();
-	bool processControlEvent(uint8 cmd);
-
-	void keyOn();
-	void nextShape();
-
-	void protect();
-	void restore();
-	virtual void reset();
-
-	void fadeStep();
-
-protected:
-	void setOutputLevel(uint8 lvl);
-
-	bool control_f0_setPatch(uint8 para);
-	bool control_f1_setTotalLevel(uint8 para);
-	bool control_f4_setAlgorithm(uint8 para);
-	bool control_f9_loadCustomPatch(uint8 para);
-	bool control_fb_incOutLevel(uint8 para);
-	bool control_fc_decOutLevel(uint8 para);
-	bool control_ff_endOfTrack(uint8 para);
-
-	typedef bool (TownsPC98_MusicChannelSSG::*ControlEventFunc)(uint8 para);
-	const ControlEventFunc *controlEvents;
-};
-
-class TownsPC98_SfxChannel : public TownsPC98_MusicChannelSSG {
-public:
-	TownsPC98_SfxChannel(TownsPC98_AudioDriver *driver, uint8 regOffs,
-	                     uint8 flgs, uint8 num, uint8 key, uint8 prt, uint8 id) :
-		TownsPC98_MusicChannelSSG(driver, regOffs, flgs, num, key, prt, id) {}
-	~TownsPC98_SfxChannel() {}
-
-	void loadData(uint8 *data);
-	void reset();
-};
-
-#ifndef DISABLE_PC98_RHYTHM_CHANNEL
-class TownsPC98_MusicChannelPCM : public TownsPC98_MusicChannel {
-public:
-	TownsPC98_MusicChannelPCM(TownsPC98_AudioDriver *driver, uint8 regOffs,
-	                          uint8 flgs, uint8 num, uint8 key, uint8 prt, uint8 id);
-	~TownsPC98_MusicChannelPCM() {}
-	void init();
-
-	void loadData(uint8 *data);
-	void processEvents();
-	bool processControlEvent(uint8 cmd);
-
-private:
-	bool control_f1_prcStart(uint8 para);
-	bool control_ff_endOfTrack(uint8 para);
-
-	typedef bool (TownsPC98_MusicChannelPCM::*ControlEventFunc)(uint8 para);
-	const ControlEventFunc *controlEvents;
-};
-#endif
-
-TownsPC98_MusicChannel::TownsPC98_MusicChannel(TownsPC98_AudioDriver *driver, uint8 regOffs, uint8 flgs, uint8 num,
-        uint8 key, uint8 prt, uint8 id) : _drv(driver), _regOffset(regOffs), _flags(flgs), _chanNum(num), _keyNum(key),
-	_part(prt), _idFlag(id), controlEvents(0) {
-
-	_ticksLeft = _algorithm = _instr = _totalLevel = _frqBlockMSB = _keyOffTime = 0;
-	_ssgStartLvl = _ssgTl = _ssgStep = _ssgTicksLeft = _ssgTargetLvl = _block = 0;
-	_vbrInitDelayHi = _vbrInitDelayLo = _vbrDuration = _vbrCurDelay = _vbrDurLeft = 0;
-	_frqLSB = 0;
-	_hold = false;
-	_dataPtr = 0;
-	_vbrModInitVal = _vbrModCurVal = 0;
-	_frequency = 0;
-}
-
-TownsPC98_MusicChannel::~TownsPC98_MusicChannel() {
-}
-
-void TownsPC98_MusicChannel::init() {
-#define Control(x) &TownsPC98_MusicChannel::control_##x
-	static const ControlEventFunc ctrlEvents[] = {
-		Control(f0_setPatch),
-		Control(f1_presetOutputLevel),
-		Control(f2_setKeyOffTime),
-		Control(f3_setFreqLSB),
-		Control(f4_setOutputLevel),
-		Control(f5_setTempo),
-		Control(f6_repeatSection),
-		Control(f7_setupVibrato),
-		Control(f8_toggleVibrato),
-		Control(dummy),
-		Control(fa_writeReg),
-		Control(fb_incOutLevel),
-		Control(fc_decOutLevel),
-		Control(fd_jump),
-		Control(dummy),
-		Control(ff_endOfTrack)
-	};
-#undef Control
-
-	controlEvents = ctrlEvents;
-}
-
-void TownsPC98_MusicChannel::keyOff() {
-	// all operators off
-	uint8 value = _keyNum & 0x0f;
-	if (_part)
-		value |= 4;
-	uint8 regAddress = 0x28;
-	_drv->writeReg(0, regAddress, value);
-	_flags |= CHS_KEYOFF;
-}
-
-void TownsPC98_MusicChannel::keyOn() {
-	// all operators on
-	uint8 value = _keyNum | 0xf0;
-	if (_part)
-		value |= 4;
-	uint8 regAddress = 0x28;
-	_drv->writeReg(0, regAddress, value);
-}
-
-void TownsPC98_MusicChannel::loadData(uint8 *data) {
-	_flags = (_flags & ~CHS_EOT) | CHS_ALLOFF;
-	_ticksLeft = 1;
-	_dataPtr = data;
-	_totalLevel = 0x7F;
-
-	uint8 *tmp = _dataPtr;
-	for (bool loop = true; loop;) {
-		uint8 cmd = *tmp++;
-		if (cmd < 0xf0) {
-			tmp++;
-		} else if (cmd == 0xff) {
-			if (READ_LE_UINT16(tmp)) {
-				_drv->_looping |= _idFlag;
-				tmp += _drv->_opnFxCmdLen[cmd - 240];
-			} else
-				loop = false;
-		} else if (cmd == 0xf6) {
-			// reset repeat section countdown
-			tmp[0] = tmp[1];
-			tmp += 4;
-		} else {
-			tmp += _drv->_opnFxCmdLen[cmd - 240];
-		}
-	}
-}
-
-void TownsPC98_MusicChannel::processEvents() {
-	if (_flags & CHS_EOT)
-		return;
-
-	if (!_hold && _ticksLeft == _keyOffTime)
-		keyOff();
-
-	if (--_ticksLeft)
-		return;
-
-	if (!_hold)
-		keyOff();
-
-	uint8 cmd = 0;
-	bool loop = true;
-
-	while (loop) {
-		cmd = *_dataPtr++;
-		if (cmd < 0xf0)
-			loop = false;
-		else if (!processControlEvent(cmd))
-			return;
-	}
-
-	uint8 para = *_dataPtr++;
-
-	if (cmd == 0x80) {
-		keyOff();
-		_hold = false;
-	} else {
-		keyOn();
-
-		if (_hold == false || cmd != _frqBlockMSB)
-			_flags |= CHS_RECALCFREQ;
-
-		_hold = (para & 0x80) ? true : false;
-		_frqBlockMSB = cmd;
-	}
-
-	_ticksLeft = para & 0x7f;
-}
-
-void TownsPC98_MusicChannel::processFrequency() {
-	if (_flags & CHS_RECALCFREQ) {
-
-		_frequency = (READ_LE_UINT16(&_drv->_opnFreqTable[(_frqBlockMSB & 0x0f) << 1]) + _frqLSB) | (((_frqBlockMSB & 0x70) >> 1) << 8);
-
-		_drv->writeReg(_part, _regOffset + 0xa4, (_frequency >> 8));
-		_drv->writeReg(_part, _regOffset + 0xa0, (_frequency & 0xff));
-
-		setupVibrato();
-	}
-
-	if (!(_flags & CHS_VBROFF)) {
-		if (!processVibrato())
-			return;
-
-		_drv->writeReg(_part, _regOffset + 0xa4, (_frequency >> 8));
-		_drv->writeReg(_part, _regOffset + 0xa0, (_frequency & 0xff));
-	}
-}
-
-void TownsPC98_MusicChannel::setupVibrato() {
-	_vbrCurDelay = _vbrInitDelayHi;
-	if (_flags & CHS_KEYOFF) {
-		_vbrModCurVal = _vbrModInitVal;
-		_vbrCurDelay += _vbrInitDelayLo;
-	}
-	_vbrDurLeft = (_vbrDuration >> 1);
-	_flags &= ~(CHS_KEYOFF | CHS_RECALCFREQ);
-}
-
-bool TownsPC98_MusicChannel::processVibrato() {
-	if (--_vbrCurDelay)
-		return false;
-
-	_vbrCurDelay = _vbrInitDelayHi;
-	_frequency += _vbrModCurVal;
-
-	if (!--_vbrDurLeft) {
-		_vbrDurLeft = _vbrDuration;
-		_vbrModCurVal = -_vbrModCurVal;
-	}
-
-	return true;
-}
-
-bool TownsPC98_MusicChannel::processControlEvent(uint8 cmd) {
-	uint8 para = *_dataPtr++;
-	return (this->*controlEvents[cmd & 0x0f])(para);
-}
-
-void TownsPC98_MusicChannel::setOutputLevel() {
-	uint8 outopr = _drv->_opnCarrier[_algorithm];
-	uint8 reg = 0x40 + _regOffset;
-
-	for (int i = 0; i < 4; i++) {
-		if (outopr & 1)
-			_drv->writeReg(_part, reg, _totalLevel);
-		outopr >>= 1;
-		reg += 4;
-	}
-}
-
-void TownsPC98_MusicChannel::fadeStep() {
-	_totalLevel += 3;
-	if (_totalLevel > 0x7f)
-		_totalLevel = 0x7f;
-	setOutputLevel();
-}
-
-void TownsPC98_MusicChannel::reset() {
-	_hold = false;
-	_keyOffTime = 0;
-	_ticksLeft = 1;
-
-	_flags = (_flags & ~CHS_EOT) | CHS_ALLOFF;
-
-	_totalLevel = 0;
-	_algorithm = 0;
-	_flags = CHS_EOT;
-	_algorithm = 0;
-
-	_block = 0;
-	_frequency = 0;
-	_frqBlockMSB = 0;
-	_frqLSB = 0;
-
-	_ssgTl = 0;
-	_ssgStartLvl = 0;
-	_ssgTargetLvl = 0;
-	_ssgStep = 0;
-	_ssgTicksLeft = 0;
-
-	_vbrInitDelayHi = 0;
-	_vbrInitDelayLo = 0;
-	_vbrModInitVal = 0;
-	_vbrDuration = 0;
-	_vbrCurDelay = 0;
-	_vbrModCurVal = 0;
-	_vbrDurLeft = 0;
-}
-
-bool TownsPC98_MusicChannel::control_f0_setPatch(uint8 para) {
-	_instr = para;
-	uint8 reg = _regOffset + 0x80;
-
-	for (int i = 0; i < 4; i++) {
-		// set release rate for each operator
-		_drv->writeReg(_part, reg, 0x0f);
-		reg += 4;
-	}
-
-	const uint8 *tptr = _drv->_patches + ((uint32)_instr << 5);
-	reg = _regOffset + 0x30;
-
-	// write registers 0x30 to 0x8f
-	for (int i = 0; i < 6; i++) {
-		_drv->writeReg(_part, reg, tptr[0]);
-		reg += 4;
-		_drv->writeReg(_part, reg, tptr[2]);
-		reg += 4;
-		_drv->writeReg(_part, reg, tptr[1]);
-		reg += 4;
-		_drv->writeReg(_part, reg, tptr[3]);
-		reg += 4;
-		tptr += 4;
-	}
-
-	reg = _regOffset + 0xB0;
-	_algorithm = tptr[0] & 7;
-	// set feedback and algorithm
-	_drv->writeReg(_part, reg, tptr[0]);
-
-	setOutputLevel();
-	return true;
-}
-
-bool TownsPC98_MusicChannel::control_f1_presetOutputLevel(uint8 para) {
-	if (_drv->_fading)
-		return true;
-
-	_totalLevel = _drv->_opnLvlPresets[para];
-	setOutputLevel();
-	return true;
-}
-
-bool TownsPC98_MusicChannel::control_f2_setKeyOffTime(uint8 para) {
-	_keyOffTime = para;
-	return true;
-}
-
-bool TownsPC98_MusicChannel::control_f3_setFreqLSB(uint8 para) {
-	_frqLSB = (int8) para;
-	return true;
-}
-
-bool TownsPC98_MusicChannel::control_f4_setOutputLevel(uint8 para) {
-	if (_drv->_fading)
-		return true;
-
-	_totalLevel = para;
-	setOutputLevel();
-	return true;
-}
-
-bool TownsPC98_MusicChannel::control_f5_setTempo(uint8 para) {
-	_drv->setMusicTempo(para);
-	return true;
-}
-
-bool TownsPC98_MusicChannel::control_f6_repeatSection(uint8 para) {
-	_dataPtr--;
-	_dataPtr[0]--;
-
-	if (*_dataPtr) {
-		// repeat section until counter has reached zero
-		_dataPtr = _drv->_trackPtr + READ_LE_UINT16(_dataPtr + 2);
-	} else {
-		// reset counter, advance to next section
-		_dataPtr[0] = _dataPtr[1];
-		_dataPtr += 4;
-	}
-	return true;
-}
-
-bool TownsPC98_MusicChannel::control_f7_setupVibrato(uint8 para) {
-	_vbrInitDelayHi = _dataPtr[0];
-	_vbrInitDelayLo = para;
-	_vbrModInitVal = (int16) READ_LE_UINT16(_dataPtr + 1);
-	_vbrDuration = _dataPtr[3];
-	_dataPtr += 4;
-	_flags = (_flags & ~CHS_VBROFF) | CHS_KEYOFF | CHS_RECALCFREQ;
-	return true;
-}
-
-bool TownsPC98_MusicChannel::control_f8_toggleVibrato(uint8 para) {
-	if (para == 0x10) {
-		if (*_dataPtr++) {
-			_flags = (_flags & ~CHS_VBROFF) | CHS_KEYOFF;
-		} else {
-			_flags |= CHS_VBROFF;
-		}
-	} else {
-		/* NOT IMPLEMENTED
-		uint8 skipChannels = para / 36;
-		uint8 entry = para % 36;
-		TownsPC98_AudioDriver::TownsPC98_MusicChannel *t = &chan[skipChannels];
-
-		t->unnamedEntries[entry] = *_dataPtr++;*/
-	}
-	return true;
-}
-
-bool TownsPC98_MusicChannel::control_fa_writeReg(uint8 para) {
-	_drv->writeReg(_part, para, *_dataPtr++);
-	return true;
-}
-
-bool TownsPC98_MusicChannel::control_fb_incOutLevel(uint8 para) {
-	_dataPtr--;
-	if (_drv->_fading)
-		return true;
-
-	uint8 val = (_totalLevel + 3);
-	if (val > 0x7f)
-		val = 0x7f;
-
-	_totalLevel = val;
-	setOutputLevel();
-	return true;
-}
-
-bool TownsPC98_MusicChannel::control_fc_decOutLevel(uint8 para) {
-	_dataPtr--;
-	if (_drv->_fading)
-		return true;
-
-	int8 val = (int8)(_totalLevel - 3);
-	if (val < 0)
-		val = 0;
-
-	_totalLevel = (uint8) val;
-	setOutputLevel();
-	return true;
-}
-
-bool TownsPC98_MusicChannel::control_fd_jump(uint8 para) {
-	uint8 *tmp = _drv->_trackPtr + READ_LE_UINT16(_dataPtr - 1);
-	_dataPtr = (tmp[1] == 1) ? tmp : (_dataPtr + 1);
-	return true;
-}
-
-bool TownsPC98_MusicChannel::control_dummy(uint8 para) {
-	_dataPtr--;
-	return true;
-}
-
-bool TownsPC98_MusicChannel::control_ff_endOfTrack(uint8 para) {
-	uint16 val = READ_LE_UINT16(--_dataPtr);
-	if (val) {
-		// loop
-		_dataPtr = _drv->_trackPtr + val;
-		return true;
-	} else {
-		// quit parsing for active channel
-		--_dataPtr;
-		_flags |= CHS_EOT;
-		_drv->_finishedChannelsFlag |= _idFlag;
-		keyOff();
-		return false;
-	}
-}
-
-TownsPC98_MusicChannelSSG::TownsPC98_MusicChannelSSG(TownsPC98_AudioDriver *driver, uint8 regOffs,
-        uint8 flgs, uint8 num, uint8 key, uint8 prt, uint8 id) :
-	TownsPC98_MusicChannel(driver, regOffs, flgs, num, key, prt, id), controlEvents(0) {
-}
-
-void TownsPC98_MusicChannelSSG::init() {
-	_algorithm = 0x80;
-
-#define Control(x) &TownsPC98_MusicChannelSSG::control_##x
-	static const ControlEventFunc ctrlEventsSSG[] = {
-		Control(f0_setPatch),
-		Control(f1_setTotalLevel),
-		Control(f2_setKeyOffTime),
-		Control(f3_setFreqLSB),
-		Control(f4_setAlgorithm),
-		Control(f5_setTempo),
-		Control(f6_repeatSection),
-		Control(f7_setupVibrato),
-		Control(f8_toggleVibrato),
-		Control(f9_loadCustomPatch),
-		Control(fa_writeReg),
-		Control(fb_incOutLevel),
-		Control(fc_decOutLevel),
-		Control(fd_jump),
-		Control(dummy),
-		Control(ff_endOfTrack)
-	};
-#undef Control
-
-	controlEvents = ctrlEventsSSG;
-}
-
-void TownsPC98_MusicChannelSSG::processEvents() {
-	if (_flags & CHS_EOT)
-		return;
-
-	_drv->toggleRegProtection(_flags & CHS_PROTECT ? true : false);
-
-	if (!_hold && _ticksLeft == _keyOffTime)
-		nextShape();
-
-	if (!--_ticksLeft) {
-
-		uint8 cmd = 0;
-		bool loop = true;
-
-		while (loop) {
-			cmd = *_dataPtr++;
-			if (cmd < 0xf0)
-				loop = false;
-			else if (!processControlEvent(cmd))
-				return;
-		}
-
-		uint8 para = *_dataPtr++;
-
-		if (cmd == 0x80) {
-			nextShape();
-			_hold = false;
-		} else {
-			if (!_hold) {
-				_instr &= 0xf0;
-				_ssgStep = _drv->_ssgPatches[_instr];
-				_ssgTicksLeft = _drv->_ssgPatches[_instr + 1] & 0x7f;
-				_ssgTargetLvl = _drv->_ssgPatches[_instr + 2];
-				_ssgStartLvl = _drv->_ssgPatches[_instr + 3];
-				_flags = (_flags & ~CHS_SSGOFF) | CHS_KEYOFF;
-			}
-
-			keyOn();
-
-			if (_hold == false || cmd != _frqBlockMSB)
-				_flags |= CHS_RECALCFREQ;
-
-			_hold = (para & 0x80) ? true : false;
-			_frqBlockMSB = cmd;
-		}
-
-		_ticksLeft = para & 0x7f;
-	}
-
-	if (!(_flags & CHS_SSGOFF)) {
-		if (--_ssgTicksLeft) {
-			if (!_drv->_fading)
-				setOutputLevel(_ssgStartLvl);
-			return;
-		}
-
-		_ssgTicksLeft = _drv->_ssgPatches[_instr + 1] & 0x7f;
-
-		if (_drv->_ssgPatches[_instr + 1] & 0x80) {
-			uint8 t = _ssgStartLvl - _ssgStep;
-
-			if (_ssgStep <= _ssgStartLvl && _ssgTargetLvl < t) {
-				if (!_drv->_fading)
-					setOutputLevel(t);
-				return;
-			}
-		} else {
-			int t = _ssgStartLvl + _ssgStep;
-			uint8 p = (uint8)(t & 0xff);
-
-			if (t < 256 && _ssgTargetLvl > p) {
-				if (!_drv->_fading)
-					setOutputLevel(p);
-				return;
-			}
-		}
-
-		setOutputLevel(_ssgTargetLvl);
-		if (_ssgStartLvl && !(_instr & 8)) {
-			_instr += 4;
-			_ssgStep = _drv->_ssgPatches[_instr];
-			_ssgTicksLeft = _drv->_ssgPatches[_instr + 1] & 0x7f;
-			_ssgTargetLvl = _drv->_ssgPatches[_instr + 2];
-		} else {
-			_flags |= CHS_SSGOFF;
-			setOutputLevel(0);
-		}
-	}
-}
-
-void TownsPC98_MusicChannelSSG::processFrequency() {
-	if (_algorithm & 0x40)
-		return;
-
-	if (_flags & CHS_RECALCFREQ) {
-		_block = _frqBlockMSB >> 4;
-		_frequency = READ_LE_UINT16(&_drv->_opnFreqTableSSG[(_frqBlockMSB & 0x0f) << 1]) + _frqLSB;
-
-		uint16 f = _frequency >> _block;
-		_drv->writeReg(_part, _regOffset << 1, f & 0xff);
-		_drv->writeReg(_part, (_regOffset << 1) + 1, f >> 8);
-
-		setupVibrato();
-	}
-
-	if (!(_flags & (CHS_EOT | CHS_VBROFF | CHS_SSGOFF))) {
-		if (!processVibrato())
-			return;
-
-		uint16 f = _frequency >> _block;
-		_drv->writeReg(_part, _regOffset << 1, f & 0xff);
-		_drv->writeReg(_part, (_regOffset << 1) + 1, f >> 8);
-	}
-}
-
-bool TownsPC98_MusicChannelSSG::processControlEvent(uint8 cmd) {
-	uint8 para = *_dataPtr++;
-	return (this->*controlEvents[cmd & 0x0f])(para);
-}
-
-void TownsPC98_MusicChannelSSG::nextShape() {
-	_instr = (_instr & 0xf0) + 0x0c;
-	_ssgStep = _drv->_ssgPatches[_instr];
-	_ssgTicksLeft = _drv->_ssgPatches[_instr + 1] & 0x7f;
-	_ssgTargetLvl = _drv->_ssgPatches[_instr + 2];
-}
-
-void TownsPC98_MusicChannelSSG::keyOn() {
-	uint8 c = 0x7b;
-	uint8 t = (_algorithm & 0xC0) << 1;
-	if (_algorithm & 0x80)
-		t |= 4;
-
-	c = (c << (_regOffset + 1)) | (c >> (7 - _regOffset));
-	t = (t << (_regOffset + 1)) | (t >> (7 - _regOffset));
-
-	if (!(_algorithm & 0x80))
-		_drv->writeReg(_part, 6, _algorithm & 0x7f);
-
-	uint8 e = (_drv->readSSGStatus() & c) | t;
-	_drv->writeReg(_part, 7, e);
-}
-
-void TownsPC98_MusicChannelSSG::protect() {
-	_flags |= CHS_PROTECT;
-}
-
-void TownsPC98_MusicChannelSSG::restore() {
-	_flags &= ~CHS_PROTECT;
-	keyOn();
-	_drv->writeReg(_part, 8 + _regOffset, _ssgTl);
-	uint16 f = _frequency >> _block;
-	_drv->writeReg(_part, _regOffset << 1, f & 0xff);
-	_drv->writeReg(_part, (_regOffset << 1) + 1, f >> 8);
-}
-
-void TownsPC98_MusicChannelSSG::loadData(uint8 *data) {
-	_drv->toggleRegProtection(_flags & CHS_PROTECT ? true : false);
-	TownsPC98_MusicChannel::loadData(data);
-	setOutputLevel(0);
-	_algorithm = 0x80;
-}
-
-void TownsPC98_MusicChannelSSG::setOutputLevel(uint8 lvl) {
-	_ssgStartLvl = lvl;
-	uint16 newTl = (((uint16)_totalLevel + 1) * (uint16)lvl) >> 8;
-	if (newTl == _ssgTl)
-		return;
-	_ssgTl = newTl;
-	_drv->writeReg(_part, 8 + _regOffset, _ssgTl);
-}
-
-void TownsPC98_MusicChannelSSG::reset() {
-	TownsPC98_MusicChannel::reset();
-
-	// Unlike the original we restore the default patch data. This fixes a bug
-	// where certain sound effects would bring each other out of tune (e.g. the
-	// dragon's fire in Darm's house in Kyra 1 would sound different each time
-	// you triggered another sfx by dropping an item etc.)
-	uint8 i = (10 + _regOffset) << 4;
-	const uint8 *src = &_drv->_drvTables[156];
-	_drv->_ssgPatches[i] = src[i];
-	_drv->_ssgPatches[i + 3] = src[i + 3];
-	_drv->_ssgPatches[i + 4] = src[i + 4];
-	_drv->_ssgPatches[i + 6] = src[i + 6];
-	_drv->_ssgPatches[i + 8] = src[i + 8];
-	_drv->_ssgPatches[i + 12] = src[i + 12];
-}
-
-void TownsPC98_MusicChannelSSG::fadeStep() {
-	_totalLevel--;
-	if ((int8)_totalLevel < 0)
-		_totalLevel = 0;
-	setOutputLevel(_ssgStartLvl);
-}
-
-bool TownsPC98_MusicChannelSSG::control_f0_setPatch(uint8 para) {
-	_instr = para << 4;
-	para = (para >> 3) & 0x1e;
-	if (para)
-		return control_f4_setAlgorithm(para | 0x40);
-	return true;
-}
-
-bool TownsPC98_MusicChannelSSG::control_f1_setTotalLevel(uint8 para) {
-	if (!_drv->_fading)
-		_totalLevel = para;
-	return true;
-}
-
-bool TownsPC98_MusicChannelSSG::control_f4_setAlgorithm(uint8 para) {
-	_algorithm = para;
-	return true;
-}
-
-bool TownsPC98_MusicChannelSSG::control_f9_loadCustomPatch(uint8 para) {
-	_instr = (_drv->_sfxOffs + 10 + _regOffset) << 4;
-	_drv->_ssgPatches[_instr] = *_dataPtr++;
-	_drv->_ssgPatches[_instr + 3] = para;
-	_drv->_ssgPatches[_instr + 4] = *_dataPtr++;
-	_drv->_ssgPatches[_instr + 6] = *_dataPtr++;
-	_drv->_ssgPatches[_instr + 8] = *_dataPtr++;
-	_drv->_ssgPatches[_instr + 12] = *_dataPtr++;
-	return true;
-}
-
-bool TownsPC98_MusicChannelSSG::control_fb_incOutLevel(uint8 para) {
-	_dataPtr--;
-	if (_drv->_fading)
-		return true;
-
-	_totalLevel--;
-	if ((int8)_totalLevel < 0)
-		_totalLevel = 0;
-
-	return true;
-}
-
-bool TownsPC98_MusicChannelSSG::control_fc_decOutLevel(uint8 para) {
-	_dataPtr--;
-	if (_drv->_fading)
-		return true;
-
-	if (_totalLevel + 1 < 0x10)
-		_totalLevel++;
-
-	return true;
-}
-
-bool TownsPC98_MusicChannelSSG::control_ff_endOfTrack(uint8 para) {
-	if (!_drv->_sfxOffs) {
-		uint16 val = READ_LE_UINT16(--_dataPtr);
-		if (val) {
-			// loop
-			_dataPtr = _drv->_trackPtr + val;
-			return true;
-		} else {
-			// stop parsing
-			if (!_drv->_fading)
-				setOutputLevel(0);
-			--_dataPtr;
-			_flags |= CHS_EOT;
-			_drv->_finishedSSGFlag |= _idFlag;
-		}
-	} else {
-		// end of sfx track - restore ssg music channel
-		_flags |= CHS_EOT;
-		_drv->_finishedSfxFlag |= _idFlag;
-		_drv->_ssgChannels[_chanNum]->restore();
-	}
-
-	return false;
-}
-
-void TownsPC98_SfxChannel::loadData(uint8 *data) {
-	_flags = CHS_ALLOFF;
-	_ticksLeft = 1;
-	_dataPtr = data;
-	_ssgTl = 0xff;
-	_algorithm = 0x80;
-
-	uint8 *tmp = _dataPtr;
-	for (bool loop = true; loop;) {
-		uint8 cmd = *tmp++;
-		if (cmd < 0xf0) {
-			tmp++;
-		} else if (cmd == 0xff) {
-			loop = false;
-		} else if (cmd == 0xf6) {
-			// reset repeat section countdown
-			tmp[0] = tmp[1];
-			tmp += 4;
-		} else {
-			tmp += _drv->_opnFxCmdLen[cmd - 240];
-		}
-	}
-}
-
-void TownsPC98_SfxChannel::reset() {
-	TownsPC98_MusicChannel::reset();
-
-	// Unlike the original we restore the default patch data. This fixes a bug
-	// where certain sound effects would bring each other out of tune (e.g. the
-	// dragon's fire in Darm's house in Kyra 1 would sound different each time
-	// you triggered another sfx by dropping an item etc.)
-	uint8 i = (13 + _regOffset) << 4;
-	const uint8 *src = &_drv->_drvTables[156];
-	_drv->_ssgPatches[i] = src[i];
-	_drv->_ssgPatches[i + 3] = src[i + 3];
-	_drv->_ssgPatches[i + 4] = src[i + 4];
-	_drv->_ssgPatches[i + 6] = src[i + 6];
-	_drv->_ssgPatches[i + 8] = src[i + 8];
-	_drv->_ssgPatches[i + 12] = src[i + 12];
-}
-
-#ifndef DISABLE_PC98_RHYTHM_CHANNEL
-TownsPC98_MusicChannelPCM::TownsPC98_MusicChannelPCM(TownsPC98_AudioDriver *driver, uint8 regOffs,
-        uint8 flgs, uint8 num, uint8 key, uint8 prt, uint8 id) :
-	TownsPC98_MusicChannel(driver, regOffs, flgs, num, key, prt, id), controlEvents(0) {
-}
-
-void TownsPC98_MusicChannelPCM::init() {
-	_algorithm = 0x80;
-
-#define Control(x) &TownsPC98_MusicChannelPCM::control_##x
-	static const ControlEventFunc ctrlEventsPCM[] = {
-		Control(dummy),
-		Control(f1_prcStart),
-		Control(dummy),
-		Control(dummy),
-		Control(dummy),
-		Control(dummy),
-		Control(f6_repeatSection),
-		Control(dummy),
-		Control(dummy),
-		Control(dummy),
-		Control(fa_writeReg),
-		Control(dummy),
-		Control(dummy),
-		Control(dummy),
-		Control(dummy),
-		Control(ff_endOfTrack)
-	};
-#undef Control
-
-	controlEvents = ctrlEventsPCM;
-}
-
-void TownsPC98_MusicChannelPCM::loadData(uint8 *data) {
-	_flags = (_flags & ~CHS_EOT) | CHS_ALLOFF;
-	_ticksLeft = 1;
-	_dataPtr = data;
-	_totalLevel = 0x7F;
-}
-
-void TownsPC98_MusicChannelPCM::processEvents()  {
-	if (_flags & CHS_EOT)
-		return;
-
-	if (--_ticksLeft)
-		return;
-
-	uint8 cmd = 0;
-	bool loop = true;
-
-	while (loop) {
-		cmd = *_dataPtr++;
-		if (cmd == 0x80) {
-			loop = false;
-		} else if (cmd < 0xf0) {
-			_drv->writeReg(_part, 0x10, cmd);
-		} else if (!processControlEvent(cmd)) {
-			return;
-		}
-	}
-
-	_ticksLeft = *_dataPtr++;
-}
-
-bool TownsPC98_MusicChannelPCM::processControlEvent(uint8 cmd) {
-	uint8 para = *_dataPtr++;
-	return (this->*controlEvents[cmd & 0x0f])(para);
-}
-
-bool TownsPC98_MusicChannelPCM::control_f1_prcStart(uint8 para) {
-	_totalLevel = para;
-	_drv->writeReg(_part, 0x11, para);
-	return true;
-}
-
-bool TownsPC98_MusicChannelPCM::control_ff_endOfTrack(uint8 para) {
-	uint16 val = READ_LE_UINT16(--_dataPtr);
-	if (val) {
-		// loop
-		_dataPtr = _drv->_trackPtr + val;
-		return true;
-	} else {
-		// quit parsing for active channel
-		--_dataPtr;
-		_flags |= CHS_EOT;
-		_drv->_finishedRhythmFlag |= _idFlag;
-		return false;
-	}
-}
-#endif // DISABLE_PC98_RHYTHM_CHANNEL
-
-TownsPC98_AudioDriver::TownsPC98_AudioDriver(Audio::Mixer *mixer, EmuType type) : TownsPC98_FmSynth(mixer, type),
-	_channels(0), _ssgChannels(0), _sfxChannels(0),
-#ifndef DISABLE_PC98_RHYTHM_CHANNEL
-	_rhythmChannel(0),
-#endif
-	_trackPtr(0), _sfxData(0), _sfxOffs(0), _ssgPatches(0),
-	_patches(0), _sfxBuffer(0), _musicBuffer(0),
-
-	_opnCarrier(_drvTables + 76), _opnFreqTable(_drvTables + 108), _opnFreqTableSSG(_drvTables + 132),
-	_opnFxCmdLen(_drvTables + 36), _opnLvlPresets(_drvTables + (type == kTypeTowns ? 52 : 84)),
-
-	_updateChannelsFlag(type == kType26 ? 0x07 : 0x3F), _finishedChannelsFlag(0),
-	_updateSSGFlag(type == kTypeTowns ? 0x00 : 0x07), _finishedSSGFlag(0),
-	_updateRhythmFlag(type == kType86 ?
-#ifndef DISABLE_PC98_RHYTHM_CHANNEL
-	0x01
-#else
-	0x00
-#endif
-	: 0x00), _finishedRhythmFlag(0),
-	_updateSfxFlag(0), _finishedSfxFlag(0),
-
-	_musicTickCounter(0),
-
-	_musicVolume(255), _sfxVolume(255),
-
-	_musicPlaying(false), _sfxPlaying(false), _fading(false), _looping(0), _ready(false) {
-
-	_sfxOffsets[0] = _sfxOffsets[1] = 0;
-}
-
-TownsPC98_AudioDriver::~TownsPC98_AudioDriver() {
-	_ready = false;
-	deinit();
-
-	if (_channels) {
-		for (int i = 0; i < _numChan; i++)
-			delete _channels[i];
-		delete[] _channels;
-	}
-
-	if (_ssgChannels) {
-		for (int i = 0; i < _numSSG; i++)
-			delete _ssgChannels[i];
-		delete[] _ssgChannels;
-	}
-
-	if (_sfxChannels) {
-		for (int i = 0; i < 2; i++)
-			delete _sfxChannels[i];
-		delete[] _sfxChannels;
-	}
-#ifndef DISABLE_PC98_RHYTHM_CHANNEL
-	delete _rhythmChannel;
-#endif
-
-	delete[] _ssgPatches;
-}
-
-bool TownsPC98_AudioDriver::init() {
-	if (_ready) {
-		reset();
-		return true;
-	}
-
-	TownsPC98_FmSynth::init();
-
-	setVolumeChannelMasks(-1, 0);
-
-	_channels = new TownsPC98_MusicChannel *[_numChan];
-	for (int i = 0; i < _numChan; i++) {
-		int ii = i * 6;
-		_channels[i] = new TownsPC98_MusicChannel(this, _drvTables[ii], _drvTables[ii + 1],
-		        _drvTables[ii + 2], _drvTables[ii + 3], _drvTables[ii + 4], _drvTables[ii + 5]);
-		_channels[i]->init();
-	}
-
-	if (_numSSG) {
-		_ssgPatches = new uint8[256];
-		memcpy(_ssgPatches, _drvTables + 156, 256);
-
-		_ssgChannels = new TownsPC98_MusicChannelSSG *[_numSSG];
-		for (int i = 0; i < _numSSG; i++) {
-			int ii = i * 6;
-			_ssgChannels[i] = new TownsPC98_MusicChannelSSG(this, _drvTables[ii], _drvTables[ii + 1],
-			        _drvTables[ii + 2], _drvTables[ii + 3], _drvTables[ii + 4], _drvTables[ii + 5]);
-			_ssgChannels[i]->init();
-		}
-
-		_sfxChannels = new TownsPC98_SfxChannel *[2];
-		for (int i = 0; i < 2; i++) {
-			int ii = (i + 1) * 6;
-			_sfxChannels[i] = new TownsPC98_SfxChannel(this, _drvTables[ii], _drvTables[ii + 1],
-			        _drvTables[ii + 2], _drvTables[ii + 3], _drvTables[ii + 4], _drvTables[ii + 5]);
-			_sfxChannels[i]->init();
-		}
-	}
-
-#ifndef DISABLE_PC98_RHYTHM_CHANNEL
-	if (_hasPercussion) {
-		_rhythmChannel = new TownsPC98_MusicChannelPCM(this, 0, 0, 0, 0, 0, 1);
-		_rhythmChannel->init();
-	}
-#endif
-
-	setMusicTempo(84);
-	setSfxTempo(654);
-
-	_ready = true;
-
-	return true;
-}
-
-void TownsPC98_AudioDriver::loadMusicData(uint8 *data, bool loadPaused) {
-	if (!_ready) {
-		warning("TownsPC98_AudioDriver: Driver must be initialized before loading data");
-		return;
-	}
-
-	if (!data) {
-		warning("TownsPC98_AudioDriver: Invalid music file data");
-		return;
-	}
-
-	reset();
-
-	Common::StackLock lock(_mutex);
-	uint8 *src_a = _trackPtr = _musicBuffer = data;
-
-	for (uint8 i = 0; i < 3; i++) {
-		_channels[i]->loadData(data + READ_LE_UINT16(src_a));
-		src_a += 2;
-	}
-
-	for (int i = 0; i < _numSSG; i++) {
-		_ssgChannels[i]->loadData(data + READ_LE_UINT16(src_a));
-		src_a += 2;
-	}
-
-	for (uint8 i = 3; i < _numChan; i++) {
-		_channels[i]->loadData(data + READ_LE_UINT16(src_a));
-		src_a += 2;
-	}
-
-	if (_hasPercussion) {
-#ifndef DISABLE_PC98_RHYTHM_CHANNEL
-		_rhythmChannel->loadData(data + READ_LE_UINT16(src_a));
-#endif
-		src_a += 2;
-	}
-
-	toggleRegProtection(false);
-
-	_patches = src_a + 4;
-	_finishedChannelsFlag = _finishedSSGFlag = _finishedRhythmFlag = 0;
-
-	_musicPlaying = (loadPaused ? false : true);
-}
-
-void TownsPC98_AudioDriver::loadSoundEffectData(uint8 *data, uint8 trackNum) {
-	if (!_ready) {
-		warning("TownsPC98_AudioDriver: Driver must be initialized before loading data");
-		return;
-	}
-
-	if (!_sfxChannels) {
-		warning("TownsPC98_AudioDriver: Sound effects not supported by this configuration");
-		return;
-	}
-
-	if (!data) {
-		warning("TownsPC98_AudioDriver: Invalid sound effects file data");
-		return;
-	}
-
-	Common::StackLock lock(_mutex);
-	_sfxData = _sfxBuffer = data;
-	_sfxOffsets[0] = READ_LE_UINT16(&_sfxData[(trackNum << 2)]);
-	_sfxOffsets[1] = READ_LE_UINT16(&_sfxData[(trackNum << 2) + 2]);
-	_sfxPlaying = true;
-	_finishedSfxFlag = 0;
-}
-
-void TownsPC98_AudioDriver::reset() {
-	Common::StackLock lock(_mutex);
-
-	_musicPlaying = false;
-	_sfxPlaying = false;
-	_fading = false;
-	_looping = 0;
-	_musicTickCounter = 0;
-	_sfxData = 0;
-
-	TownsPC98_FmSynth::reset();
-
-	for (int i = 0; i < _numChan; i++)
-		_channels[i]->reset();
-	for (int i = 0; i < _numSSG; i++)
-		_ssgChannels[i]->reset();
-
-	if (_numSSG) {
-		for (int i = 0; i < 2; i++)
-			_sfxChannels[i]->reset();
-
-		memcpy(_ssgPatches, _drvTables + 156, 256);
-	}
-
-#ifndef DISABLE_PC98_RHYTHM_CHANNEL
-	if (_rhythmChannel)
-		_rhythmChannel->reset();
-#endif
-}
-
-void TownsPC98_AudioDriver::fadeStep() {
-	if (!_musicPlaying)
-		return;
-
-	Common::StackLock lock(_mutex);
-	for (int j = 0; j < _numChan; j++) {
-		if (_updateChannelsFlag & _channels[j]->_idFlag)
-			_channels[j]->fadeStep();
-	}
-
-	for (int j = 0; j < _numSSG; j++) {
-		if (_updateSSGFlag & _ssgChannels[j]->_idFlag)
-			_ssgChannels[j]->fadeStep();
-	}
-
-	if (!_fading) {
-		_fading = 19;
-#ifndef DISABLE_PC98_RHYTHM_CHANNEL
-		if (_hasPercussion) {
-			if (_updateRhythmFlag & _rhythmChannel->_idFlag)
-				_rhythmChannel->reset();
-		}
-#endif
-	} else {
-		if (!--_fading)
-			reset();
-	}
-}
-
-void TownsPC98_AudioDriver::timerCallbackB() {
-	_sfxOffs = 0;
-
-	if (_musicPlaying) {
-		_musicTickCounter++;
-
-		for (int i = 0; i < _numChan; i++) {
-			if (_updateChannelsFlag & _channels[i]->_idFlag) {
-				_channels[i]->processEvents();
-				_channels[i]->processFrequency();
-			}
-		}
-
-		for (int i = 0; i < _numSSG; i++) {
-			if (_updateSSGFlag & _ssgChannels[i]->_idFlag) {
-				_ssgChannels[i]->processEvents();
-				_ssgChannels[i]->processFrequency();
-			}
-		}
-
-#ifndef DISABLE_PC98_RHYTHM_CHANNEL
-		if (_hasPercussion)
-			if (_updateRhythmFlag & _rhythmChannel->_idFlag)
-				_rhythmChannel->processEvents();
-#endif
-	}
-
-	toggleRegProtection(false);
-
-	if (_finishedChannelsFlag == _updateChannelsFlag && _finishedSSGFlag == _updateSSGFlag && _finishedRhythmFlag == _updateRhythmFlag)
-		_musicPlaying = false;
-}
-
-void TownsPC98_AudioDriver::timerCallbackA() {
-	if (_sfxChannels && _sfxPlaying) {
-		if (_sfxData)
-			startSoundEffect();
-
-		_sfxOffs = 3;
-		_trackPtr = _sfxBuffer;
-
-		for (int i = 0; i < 2; i++) {
-			if (_updateSfxFlag & _sfxChannels[i]->_idFlag) {
-				_sfxChannels[i]->processEvents();
-				_sfxChannels[i]->processFrequency();
-			}
-		}
-
-		_trackPtr = _musicBuffer;
-	}
-
-	if (_updateSfxFlag && _finishedSfxFlag == _updateSfxFlag) {
-		_sfxPlaying = false;
-		_updateSfxFlag = 0;
-		setVolumeChannelMasks(-1, 0);
-	}
-}
-
-void TownsPC98_AudioDriver::setMusicTempo(uint8 tempo) {
-	writeReg(0, 0x26, tempo);
-	writeReg(0, 0x27, 0x33);
-}
-
-void TownsPC98_AudioDriver::setSfxTempo(uint16 tempo) {
-	writeReg(0, 0x24, tempo & 0xff);
-	writeReg(0, 0x25, tempo >> 8);
-	writeReg(0, 0x27, 0x33);
-}
-
-void TownsPC98_AudioDriver::startSoundEffect() {
-	int volFlags = 0;
-
-	for (int i = 0; i < 2; i++) {
-		if (_sfxOffsets[i]) {
-			_ssgChannels[i + 1]->protect();
-			_sfxChannels[i]->reset();
-			_sfxChannels[i]->loadData(_sfxData + _sfxOffsets[i]);
-			_updateSfxFlag |= _sfxChannels[i]->_idFlag;
-			volFlags |= (_sfxChannels[i]->_idFlag << _numChan);
-		} else {
-			_ssgChannels[i + 1]->restore();
-			_updateSfxFlag &= ~_sfxChannels[i]->_idFlag;
-		}
-	}
-
-	setVolumeChannelMasks(~volFlags, volFlags);
-	_sfxData = 0;
-}
-
-const uint8 TownsPC98_AudioDriver::_drvTables[] = {
-	//  channel presets
-	0x00, 0x80, 0x00, 0x00, 0x00, 0x01,
-	0x01, 0x80, 0x01, 0x01, 0x00, 0x02,
-	0x02, 0x80, 0x02, 0x02, 0x00, 0x04,
-	0x00, 0x80, 0x03, 0x04, 0x01, 0x08,
-	0x01, 0x80, 0x04, 0x05, 0x01, 0x10,
-	0x02, 0x80, 0x05, 0x06, 0x01, 0x20,
-
-	//  control event size
-	0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x04, 0x05,
-	0x02, 0x06, 0x02, 0x00, 0x00, 0x02, 0x00, 0x02,
-
-	//  fmt level presets
-	0x54, 0x50, 0x4C, 0x48, 0x44, 0x40, 0x3C, 0x38,
-	0x34, 0x30, 0x2C, 0x28, 0x24, 0x20, 0x1C, 0x18,
-	0x14, 0x10, 0x0C, 0x08, 0x04, 0x90, 0x90, 0x90,
-
-	//  carriers
-	0x08, 0x08, 0x08, 0x08, 0x0C, 0x0E, 0x0E, 0x0F,
-
-	//  pc98 level presets
-	0x40, 0x3B, 0x38, 0x34, 0x30, 0x2A, 0x28, 0x25,
-	0x22, 0x20, 0x1D, 0x1A, 0x18, 0x15, 0x12, 0x10,
-	0x0D, 0x0A, 0x08, 0x05, 0x02, 0x90, 0x90, 0x90,
-
-	//  frequencies
-	0x6A, 0x02, 0x8F, 0x02, 0xB6, 0x02, 0xDF, 0x02,
-	0x0B, 0x03, 0x39, 0x03, 0x6A, 0x03, 0x9E, 0x03,
-	0xD5, 0x03, 0x10, 0x04, 0x4E, 0x04, 0x8F, 0x04,
-
-	//  ssg frequencies
-	0xE8, 0x0E, 0x12, 0x0E, 0x48, 0x0D, 0x89, 0x0C,
-	0xD5, 0x0B, 0x2B, 0x0B, 0x8A, 0x0A, 0xF3, 0x09,
-	0x64, 0x09, 0xDD, 0x08, 0x5E, 0x08, 0xE6, 0x07,
-
-	// ssg patch data
-	0x00, 0x00, 0xFF, 0xFF, 0x00, 0x81, 0x00, 0x00,
-	0x00, 0x81, 0x00, 0x00, 0xFF, 0x81, 0x00, 0x00,
-	0x00, 0x01, 0xFF, 0xFF, 0x37, 0x81, 0xC8, 0x00,
-	0x00, 0x81, 0x00, 0x00, 0x0A, 0x81, 0x00, 0x00,
-	0x00, 0x01, 0xFF, 0xFF, 0x37, 0x81, 0xC8, 0x00,
-	0x01, 0x81, 0x00, 0x00, 0x0A, 0x81, 0x00, 0x00,
-	0x00, 0x01, 0xFF, 0xFF, 0xFF, 0x81, 0xBE, 0x00,
-	0x00, 0x81, 0x00, 0x00, 0x0A, 0x81, 0x00, 0x00,
-	0x00, 0x01, 0xFF, 0xFF, 0xFF, 0x81, 0xBE, 0x00,
-	0x01, 0x81, 0x00, 0x00, 0x0A, 0x81, 0x00, 0x00,
-	0x00, 0x01, 0xFF, 0xFF, 0xFF, 0x81, 0xBE, 0x00,
-	0x04, 0x81, 0x00, 0x00, 0x0A, 0x81, 0x00, 0x00,
-	0x00, 0x01, 0xFF, 0xFF, 0xFF, 0x81, 0xBE, 0x00,
-	0x0A, 0x81, 0x00, 0x00, 0x0A, 0x81, 0x00, 0x00,
-	0x00, 0x01, 0xFF, 0xFF, 0xFF, 0x81, 0x01, 0x00,
-	0xFF, 0x81, 0x00, 0x00, 0xFF, 0x81, 0x00, 0x00,
-	0xFF, 0x01, 0xFF, 0xFF, 0xFF, 0x81, 0xFF, 0x00,
-	0x01, 0x81, 0x00, 0x00, 0x0A, 0x81, 0x00, 0x00,
-	0x64, 0x01, 0xFF, 0x64, 0xFF, 0x81, 0xFF, 0x00,
-	0x01, 0x81, 0x00, 0x00, 0x0A, 0x81, 0x00, 0x00,
-
-	0x02, 0x01, 0xFF, 0x28, 0xFF, 0x81, 0xF0, 0x00,
-	0x00, 0x81, 0x00, 0x00, 0x0A, 0x81, 0x00, 0x00,
-	0x00, 0x01, 0xFF, 0xFF, 0xFF, 0x81, 0xC8, 0x00,
-	0x01, 0x81, 0x00, 0x00, 0x28, 0x81, 0x00, 0x00,
-	0x00, 0x01, 0xFF, 0x78, 0x5F, 0x81, 0xA0, 0x00,
-	0x05, 0x81, 0x00, 0x00, 0x28, 0x81, 0x00, 0x00,
-	0x00, 0x01, 0xFF, 0xFF, 0x00, 0x81, 0x00, 0x00,
-	0x00, 0x81, 0x00, 0x00, 0xFF, 0x81, 0x00, 0x00,
-	0x00, 0x01, 0xFF, 0xFF, 0x00, 0x81, 0x00, 0x00,
-	0x00, 0x81, 0x00, 0x00, 0xFF, 0x81, 0x00, 0x00,
-	0x00, 0x01, 0xFF, 0xFF, 0x00, 0x81, 0x00, 0x00,
-	0x00, 0x81, 0x00, 0x00, 0xFF, 0x81, 0x00, 0x00
-};
-
-#undef EUPHONY_FADEOUT_TICKS
-
diff --git a/sound/softsynth/fmtowns_pc98/towns_pc98_driver.h b/sound/softsynth/fmtowns_pc98/towns_pc98_driver.h
deleted file mode 100644
index 00fcf7c5d5..0000000000
--- a/sound/softsynth/fmtowns_pc98/towns_pc98_driver.h
+++ /dev/null
@@ -1,135 +0,0 @@
-/* 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 TOWNS_PC98_AUDIODRIVER_H
-#define TOWNS_PC98_AUDIODRIVER_H
-
-#include "sound/softsynth/fmtowns_pc98/towns_pc98_fmsynth.h"
-
-class TownsPC98_MusicChannel;
-class TownsPC98_MusicChannelSSG;
-class TownsPC98_SfxChannel;
-#ifndef DISABLE_PC98_RHYTHM_CHANNEL
-class TownsPC98_MusicChannelPCM;
-#endif
-
-class TownsPC98_AudioDriver : public TownsPC98_FmSynth {
-friend class TownsPC98_MusicChannel;
-friend class TownsPC98_MusicChannelSSG;
-friend class TownsPC98_SfxChannel;
-#ifndef DISABLE_PC98_RHYTHM_CHANNEL
-friend class TownsPC98_MusicChannelPCM;
-#endif
-public:
-	TownsPC98_AudioDriver(Audio::Mixer *mixer, EmuType type);
-	~TownsPC98_AudioDriver();
-
-	void loadMusicData(uint8 *data, bool loadPaused = false);
-	void loadSoundEffectData(uint8 *data, uint8 trackNum);
-	bool init();
-	void reset();
-
-	void fadeStep();
-
-	void pause() {
-		_musicPlaying = false;
-	}
-	void cont() {
-		_musicPlaying = true;
-	}
-
-	void timerCallbackB();
-	void timerCallbackA();
-
-	bool looping() {
-		return _looping == _updateChannelsFlag ? true : false;
-	}
-	bool musicPlaying() {
-		return _musicPlaying;
-	}
-
-	void setMusicVolume(int volume) {
-		_musicVolume = volume;
-		setVolumeIntern(_musicVolume, _sfxVolume);
-	}
-	void setSoundEffectVolume(int volume) {
-		_sfxVolume = volume;
-		setVolumeIntern(_musicVolume, _sfxVolume);
-	}
-
-protected:
-	void startSoundEffect();
-
-	void setMusicTempo(uint8 tempo);
-	void setSfxTempo(uint16 tempo);
-
-	TownsPC98_MusicChannel **_channels;
-	TownsPC98_MusicChannelSSG **_ssgChannels;
-	TownsPC98_SfxChannel **_sfxChannels;
-#ifndef DISABLE_PC98_RHYTHM_CHANNEL
-	TownsPC98_MusicChannelPCM *_rhythmChannel;
-#endif
-
-	const uint8 *_opnCarrier;
-	const uint8 *_opnFreqTable;
-	const uint8 *_opnFreqTableSSG;
-	const uint8 *_opnFxCmdLen;
-	const uint8 *_opnLvlPresets;
-
-	uint8 *_musicBuffer;
-	uint8 *_sfxBuffer;
-	uint8 *_trackPtr;
-	uint8 *_patches;
-	uint8 *_ssgPatches;
-
-	uint8 _updateChannelsFlag;
-	uint8 _updateSSGFlag;
-	uint8 _updateRhythmFlag;
-	uint8 _updateSfxFlag;
-	uint8 _finishedChannelsFlag;
-	uint8 _finishedSSGFlag;
-	uint8 _finishedRhythmFlag;
-	uint8 _finishedSfxFlag;
-
-	bool _musicPlaying;
-	bool _sfxPlaying;
-	uint8 _fading;
-	uint8 _looping;
-	uint32 _musicTickCounter;
-
-	int _sfxOffs;
-	uint8 *_sfxData;
-	uint16 _sfxOffsets[2];
-
-	uint16 _musicVolume;
-	uint16 _sfxVolume;
-
-	static const uint8 _drvTables[];
-
-	bool _ready;
-};
-
-#endif
-
diff --git a/sound/softsynth/fmtowns_pc98/towns_pc98_fmsynth.cpp b/sound/softsynth/fmtowns_pc98/towns_pc98_fmsynth.cpp
deleted file mode 100644
index 62f7d39771..0000000000
--- a/sound/softsynth/fmtowns_pc98/towns_pc98_fmsynth.cpp
+++ /dev/null
@@ -1,1548 +0,0 @@
-/* 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 "sound/softsynth/fmtowns_pc98/towns_pc98_fmsynth.h"
-#include "common/endian.h"
-
-class TownsPC98_FmSynthOperator {
-public:
-	TownsPC98_FmSynthOperator(const uint32 timerbase, const uint32 rtt, const uint8 *rateTable,
-	                          const uint8 *shiftTable, const uint8 *attackDecayTable, const uint32 *frqTable,
-	                          const uint32 *sineTable, const int32 *tlevelOut, const int32 *detuneTable);
-	~TownsPC98_FmSynthOperator() {}
-
-	void keyOn();
-	void keyOff();
-	void frequency(int freq);
-	void updatePhaseIncrement();
-	void recalculateRates();
-	void generateOutput(int32 phasebuf, int32 *feedbuf, int32 &out);
-
-	void feedbackLevel(int32 level) {
-		_feedbackLevel = level ? level + 6 : 0;
-	}
-	void detune(int value) {
-		_detn = &_detnTbl[value << 5];
-	}
-	void multiple(uint32 value) {
-		_multiple = value ? (value << 1) : 1;
-	}
-	void attackRate(uint32 value) {
-		_specifiedAttackRate = value;
-	}
-	bool scaleRate(uint8 value);
-	void decayRate(uint32 value) {
-		_specifiedDecayRate = value;
-		recalculateRates();
-	}
-	void sustainRate(uint32 value) {
-		_specifiedSustainRate = value;
-		recalculateRates();
-	}
-	void sustainLevel(uint32 value) {
-		_sustainLevel = (value == 0x0f) ? 0x3e0 : value << 5;
-	}
-	void releaseRate(uint32 value) {
-		_specifiedReleaseRate = value;
-		recalculateRates();
-	}
-	void totalLevel(uint32 value) {
-		_totalLevel = value << 3;
-	}
-	void ampModulation(bool enable) {
-		_ampMod = enable;
-	}
-	void reset();
-
-protected:
-	EnvelopeState _state;
-	bool _playing;
-	uint32 _feedbackLevel;
-	uint32 _multiple;
-	uint32 _totalLevel;
-	uint8 _keyScale1;
-	uint8 _keyScale2;
-	uint32 _specifiedAttackRate;
-	uint32 _specifiedDecayRate;
-	uint32 _specifiedSustainRate;
-	uint32 _specifiedReleaseRate;
-	uint32 _tickCount;
-	uint32 _sustainLevel;
-
-	bool _ampMod;
-	uint32 _frequency;
-	uint8 _kcode;
-	uint32 _phase;
-	uint32 _phaseIncrement;
-	const int32 *_detn;
-
-	const uint8 *_rateTbl;
-	const uint8 *_rshiftTbl;
-	const uint8 *_adTbl;
-	const uint32 *_fTbl;
-	const uint32 *_sinTbl;
-	const int32 *_tLvlTbl;
-	const int32 *_detnTbl;
-
-	const uint32 _tickLength;
-	uint32 _timer;
-	const uint32 _rtt;
-	int32 _currentLevel;
-
-	struct EvpState {
-		uint8 rate;
-		uint8 shift;
-	} fs_a, fs_d, fs_s, fs_r;
-};
-
-TownsPC98_FmSynthOperator::TownsPC98_FmSynthOperator(const uint32 timerbase, const uint32 rtt,
-		const uint8 *rateTable, const uint8 *shiftTable, const uint8 *attackDecayTable,
-		const uint32 *frqTable, const uint32 *sineTable, const int32 *tlevelOut, const int32 *detuneTable) :
-	_rtt(rtt), _rateTbl(rateTable), _rshiftTbl(shiftTable), _adTbl(attackDecayTable), _fTbl(frqTable),
-	_sinTbl(sineTable), _tLvlTbl(tlevelOut), _detnTbl(detuneTable), _tickLength(timerbase * 2),
-	_specifiedAttackRate(0), _specifiedDecayRate(0), _specifiedReleaseRate(0), _specifiedSustainRate(0),
-	_phase(0), _state(kEnvReady), _playing(false), _timer(0), _keyScale1(0),
-	_keyScale2(0), _currentLevel(1023), _ampMod(false), _tickCount(0) {
-
-	fs_a.rate = fs_a.shift = fs_d.rate = fs_d.shift = fs_s.rate = fs_s.shift = fs_r.rate = fs_r.shift = 0;
-
-	reset();
-}
-
-void TownsPC98_FmSynthOperator::keyOn() {
-	if (_playing)
-		return;
-
-	_playing = true;
-	_state = kEnvAttacking;
-	_phase = 0;
-}
-
-void TownsPC98_FmSynthOperator::keyOff() {
-	if (!_playing)
-		return;
-
-	_playing = false;
-	if (_state != kEnvReady)
-		_state = kEnvReleasing;
-}
-
-void TownsPC98_FmSynthOperator::frequency(int freq) {
-	uint8 block = (freq >> 11);
-	uint16 pos = (freq & 0x7ff);
-	uint8 c = pos >> 7;
-
-	_kcode = (block << 2) | ((c < 7) ? 0 : ((c > 8) ? 3 : c - 6));
-	_frequency = _fTbl[pos << 1] >> (7 - block);
-}
-
-void TownsPC98_FmSynthOperator::updatePhaseIncrement() {
-	_phaseIncrement = ((_frequency + _detn[_kcode]) * _multiple) >> 1;
-	uint8 keyscale = _kcode >> _keyScale1;
-	if (_keyScale2 != keyscale) {
-		_keyScale2 = keyscale;
-		recalculateRates();
-	}
-}
-
-void TownsPC98_FmSynthOperator::recalculateRates() {
-	int k = _keyScale2;
-	int r = _specifiedAttackRate ? (_specifiedAttackRate << 1) + 0x20 : 0;
-	fs_a.rate = ((r + k) < 94) ? _rateTbl[r + k] : 136;
-	fs_a.shift = ((r + k) < 94) ? _rshiftTbl[r + k] : 0;
-
-	r = _specifiedDecayRate ? (_specifiedDecayRate << 1) + 0x20 : 0;
-	fs_d.rate = _rateTbl[r + k];
-	fs_d.shift = _rshiftTbl[r + k];
-
-	r = _specifiedSustainRate ? (_specifiedSustainRate << 1) + 0x20 : 0;
-	fs_s.rate = _rateTbl[r + k];
-	fs_s.shift = _rshiftTbl[r + k];
-
-	r = (_specifiedReleaseRate << 2) + 0x22;
-	fs_r.rate = _rateTbl[r + k];
-	fs_r.shift = _rshiftTbl[r + k];
-}
-
-void TownsPC98_FmSynthOperator::generateOutput(int32 phasebuf, int32 *feed, int32 &out) {
-	if (_state == kEnvReady)
-		return;
-
-	_timer += _tickLength;
-	while (_timer > _rtt) {
-		_timer -= _rtt;
-		++_tickCount;
-
-		int32 levelIncrement = 0;
-		uint32 targetTime = 0;
-		int32 targetLevel = 0;
-		EnvelopeState nextState = kEnvReady;
-
-		switch (_state) {
-		case kEnvReady:
-			return;
-		case kEnvAttacking:
-			targetLevel = 0;
-			nextState = kEnvDecaying;
-			if ((_specifiedAttackRate << 1) + _keyScale2 < 64) {
-				targetTime = (1 << fs_a.shift) - 1;
-				levelIncrement = (~_currentLevel * _adTbl[fs_a.rate + ((_tickCount >> fs_a.shift) & 7)]) >> 4;
-				break;
-			} else {
-				_currentLevel = targetLevel;
-				_state = nextState;
-			}
-			// Fall through
-		case kEnvDecaying:
-			targetTime = (1 << fs_d.shift) - 1;
-			nextState = kEnvSustaining;
-			targetLevel = _sustainLevel;
-			levelIncrement = _adTbl[fs_d.rate + ((_tickCount >> fs_d.shift) & 7)];
-			break;
-		case kEnvSustaining:
-			targetTime = (1 << fs_s.shift) - 1;
-			nextState = kEnvSustaining;
-			targetLevel = 1023;
-			levelIncrement = _adTbl[fs_s.rate + ((_tickCount >> fs_s.shift) & 7)];
-			break;
-		case kEnvReleasing:
-			targetTime = (1 << fs_r.shift) - 1;
-			nextState = kEnvReady;
-			targetLevel = 1023;
-			levelIncrement = _adTbl[fs_r.rate + ((_tickCount >> fs_r.shift) & 7)];
-			break;
-		}
-
-		if (!(_tickCount & targetTime)) {
-			_currentLevel += levelIncrement;
-			if ((_state == kEnvAttacking && _currentLevel <= targetLevel) || (_state != kEnvAttacking && _currentLevel >= targetLevel)) {
-				if (_state != kEnvDecaying)
-					_currentLevel = targetLevel;
-				_state = nextState;
-			}
-		}
-	}
-
-	uint32 lvlout = _totalLevel + (uint32) _currentLevel;
-
-
-	int32 outp = 0;
-	int32 *i = &outp, *o = &outp;
-	int phaseShift = 0;
-
-	if (feed) {
-		o = &feed[0];
-		i = &feed[1];
-		phaseShift = _feedbackLevel ? ((*o + *i) << _feedbackLevel) : 0;
-		*o = *i;
-	} else {
-		phaseShift = phasebuf << 15;
-	}
-
-	if (lvlout < 832) {
-		uint32 index = (lvlout << 3) + _sinTbl[(((int32)((_phase & 0xffff0000)
-		                                        + phaseShift)) >> 16) & 0x3ff];
-		*i = ((index < 6656) ? _tLvlTbl[index] : 0);
-	} else {
-		*i = 0;
-	}
-
-	_phase += _phaseIncrement;
-	out += *o;
-}
-
-void TownsPC98_FmSynthOperator::reset() {
-	keyOff();
-	_timer = 0;
-	_keyScale2 = 0;
-	_currentLevel = 1023;
-
-	frequency(0);
-	detune(0);
-	scaleRate(0);
-	multiple(0);
-	updatePhaseIncrement();
-	attackRate(0);
-	decayRate(0);
-	releaseRate(0);
-	sustainRate(0);
-	feedbackLevel(0);
-	totalLevel(127);
-	ampModulation(false);
-}
-
-bool TownsPC98_FmSynthOperator::scaleRate(uint8 value) {
-	value = 3 - value;
-	if (_keyScale1 != value) {
-		_keyScale1 = value;
-		return true;
-	}
-
-	int k = _keyScale2;
-	int r = _specifiedAttackRate ? (_specifiedAttackRate << 1) + 0x20 : 0;
-	fs_a.rate = ((r + k) < 94) ? _rateTbl[r + k] : 136;
-	fs_a.shift = ((r + k) < 94) ? _rshiftTbl[r + k] : 0;
-	return false;
-}
-
-class TownsPC98_FmSynthSquareSineSource {
-public:
-	TownsPC98_FmSynthSquareSineSource(const uint32 timerbase, const uint32 rtt);
-	~TownsPC98_FmSynthSquareSineSource();
-
-	void init(const int *rsTable, const int *rseTable);
-	void reset();
-	void writeReg(uint8 address, uint8 value, bool force = false);
-
-	void nextTick(int32 *buffer, uint32 bufferSize);
-
-	void setVolumeIntern(int volA, int volB) {
-		_volumeA = volA;
-		_volumeB = volB;
-	}
-	void setVolumeChannelMasks(int channelMaskA, int channelMaskB) {
-		_volMaskA = channelMaskA;
-		_volMaskB = channelMaskB;
-	}
-
-	uint8 chanEnable() const {
-		return _chanEnable;
-	}
-private:
-	void updateRegs();
-
-	uint8 _updateRequestBuf[64];
-	int _updateRequest;
-	int _rand;
-
-	int8 _evpTimer;
-	uint32 _pReslt;
-	uint8 _attack;
-
-	bool _evpUpdate, _cont;
-
-	int _evpUpdateCnt;
-	uint8 _outN;
-	int _nTick;
-
-	int32 *_tlTable;
-	int32 *_tleTable;
-
-	const uint32 _tickLength;
-	uint32 _timer;
-	const uint32 _rtt;
-
-	struct Channel {
-		int tick;
-		uint8 smp;
-		uint8 out;
-
-		uint8 frqL;
-		uint8 frqH;
-		uint8 vol;
-	} _channels[3];
-
-	uint8 _noiseGenerator;
-	uint8 _chanEnable;
-
-	uint8 **_reg;
-
-	uint16 _volumeA;
-	uint16 _volumeB;
-	int _volMaskA;
-	int _volMaskB;
-
-	bool _ready;
-};
-
-#ifndef DISABLE_PC98_RHYTHM_CHANNEL
-class TownsPC98_FmSynthPercussionSource {
-public:
-	TownsPC98_FmSynthPercussionSource(const uint32 timerbase, const uint32 rtt);
-	~TownsPC98_FmSynthPercussionSource() {
-		delete[] _reg;
-	}
-
-	void init(const uint8 *instrData = 0);
-	void reset();
-	void writeReg(uint8 address, uint8 value);
-
-	void nextTick(int32 *buffer, uint32 bufferSize);
-
-	void setVolumeIntern(int volA, int volB) {
-		_volumeA = volA;
-		_volumeB = volB;
-	}
-	void setVolumeChannelMasks(int channelMaskA, int channelMaskB) {
-		_volMaskA = channelMaskA;
-		_volMaskB = channelMaskB;
-	}
-
-private:
-	struct RhtChannel {
-		const uint8 *data;
-
-		const uint8 *start;
-		const uint8 *end;
-		const uint8 *pos;
-		uint32 size;
-		bool active;
-		uint8 level;
-
-		int8 decState;
-		uint8 decStep;
-
-		int16 samples[2];
-		int out;
-
-		uint8 startPosH;
-		uint8 startPosL;
-		uint8 endPosH;
-		uint8 endPosL;
-	};
-
-	void recalcOuput(RhtChannel *ins);
-	void advanceInput(RhtChannel *ins);
-
-	RhtChannel _rhChan[6];
-
-	uint8 _totalLevel;
-
-	const uint32 _tickLength;
-	uint32 _timer;
-	const uint32 _rtt;
-
-	uint8 **_reg;
-
-	uint16 _volumeA;
-	uint16 _volumeB;
-	int _volMaskA;
-	int _volMaskB;
-
-	bool _ready;
-};
-#endif // DISABLE_PC98_RHYTHM_CHANNEL
-
-TownsPC98_FmSynthSquareSineSource::TownsPC98_FmSynthSquareSineSource(const uint32 timerbase, const uint32 rtt) : _tlTable(0),
-	_rtt(rtt), _tleTable(0), _updateRequest(-1), _tickLength(timerbase * 27), _ready(0), _reg(0), _rand(1), _outN(1),
-	_nTick(0), _evpUpdateCnt(0), _evpTimer(0x1f), _pReslt(0x1f), _attack(0), _cont(false), _evpUpdate(true),
-	_timer(0), _noiseGenerator(0), _chanEnable(0),
-	_volMaskA(0), _volMaskB(0), _volumeA(Audio::Mixer::kMaxMixerVolume), _volumeB(Audio::Mixer::kMaxMixerVolume) {
-
-	memset(_channels, 0, sizeof(_channels));
-	memset(_updateRequestBuf, 0, sizeof(_updateRequestBuf));
-	_reg = new uint8 *[11];
-
-	_reg[0] = &_channels[0].frqL;
-	_reg[1] = &_channels[0].frqH;
-	_reg[2] = &_channels[1].frqL;
-	_reg[3] = &_channels[1].frqH;
-	_reg[4] = &_channels[2].frqL;
-	_reg[5] = &_channels[2].frqH;
-	_reg[6] = &_noiseGenerator;
-	_reg[7] = &_chanEnable;
-	_reg[8] = &_channels[0].vol;
-	_reg[9] = &_channels[1].vol;
-	_reg[10] = &_channels[2].vol;
-
-	reset();
-}
-
-TownsPC98_FmSynthSquareSineSource::~TownsPC98_FmSynthSquareSineSource() {
-	delete[] _tlTable;
-	delete[] _tleTable;
-	delete[] _reg;
-}
-
-void TownsPC98_FmSynthSquareSineSource::init(const int *rsTable, const int *rseTable) {
-	if (_ready) {
-		reset();
-		return;
-	}
-
-	delete[] _tlTable;
-	delete[] _tleTable;
-	_tlTable = new int32[16];
-	_tleTable = new int32[32];
-	float a, b, d;
-	d = 801.0f;
-
-	for (int i = 0; i < 16; i++) {
-		b = 1.0f / rsTable[i];
-		a = 1.0f / d + b + 1.0f / 1000.0f;
-		float v = (b / a) * 32767.0f;
-		_tlTable[i] = (int32) v;
-
-		b = 1.0f / rseTable[i];
-		a = 1.0f / d + b + 1.0f / 1000.0f;
-		v = (b / a) * 32767.0f;
-		_tleTable[i] = (int32) v;
-	}
-
-	for (int i = 16; i < 32; i++) {
-		b = 1.0f / rseTable[i];
-		a = 1.0f / d + b + 1.0f / 1000.0f;
-		float v = (b / a) * 32767.0f;
-		_tleTable[i] = (int32) v;
-	}
-
-	_ready = true;
-}
-
-void TownsPC98_FmSynthSquareSineSource::reset() {
-	_rand = 1;
-	_outN = 1;
-	_updateRequest = -1;
-	_nTick = _evpUpdateCnt = 0;
-	_evpTimer = 0x1f;
-	_pReslt = 0x1f;
-	_attack = 0;
-	_cont = false;
-	_evpUpdate = true;
-	_timer = 0;
-
-	for (int i = 0; i < 3; i++) {
-		_channels[i].tick = 0;
-		_channels[i].smp = _channels[i].out = 0;
-	}
-
-	for (int i = 0; i < 14; i++)
-		writeReg(i, 0, true);
-
-	writeReg(7, 0xbf, true);
-}
-
-void TownsPC98_FmSynthSquareSineSource::writeReg(uint8 address, uint8 value, bool force) {
-	if (!_ready)
-		return;
-
-	if (address > 10 || *_reg[address] == value) {
-		if ((address == 11 || address == 12 || address == 13) && value)
-			warning("TownsPC98_FmSynthSquareSineSource: unsupported reg address: %d", address);
-		return;
-	}
-
-	if (!force) {
-		if (_updateRequest >= 63) {
-			warning("TownsPC98_FmSynthSquareSineSource: event buffer overflow");
-			_updateRequest = -1;
-		}
-		_updateRequestBuf[++_updateRequest] = value;
-		_updateRequestBuf[++_updateRequest] = address;
-		return;
-	}
-
-	*_reg[address] = value;
-}
-
-void TownsPC98_FmSynthSquareSineSource::nextTick(int32 *buffer, uint32 bufferSize) {
-	if (!_ready)
-		return;
-
-	for (uint32 i = 0; i < bufferSize; i++) {
-		_timer += _tickLength;
-		while (_timer > _rtt) {
-			_timer -= _rtt;
-
-			if (++_nTick >= (_noiseGenerator & 0x1f)) {
-				if ((_rand + 1) & 2)
-					_outN ^= 1;
-
-				_rand = (((_rand & 1) ^ ((_rand >> 3) & 1)) << 16) | (_rand >> 1);
-				_nTick = 0;
-			}
-
-			for (int ii = 0; ii < 3; ii++) {
-				if (++_channels[ii].tick >= (((_channels[ii].frqH & 0x0f) << 8) | _channels[ii].frqL)) {
-					_channels[ii].tick = 0;
-					_channels[ii].smp ^= 1;
-				}
-				_channels[ii].out = (_channels[ii].smp | ((_chanEnable >> ii) & 1)) & (_outN | ((_chanEnable >> (ii + 3)) & 1));
-			}
-
-			if (_evpUpdate) {
-				if (++_evpUpdateCnt >= 0) {
-					_evpUpdateCnt = 0;
-
-					if (--_evpTimer < 0) {
-						if (_cont) {
-							_evpTimer &= 0x1f;
-						} else {
-							_evpUpdate = false;
-							_evpTimer = 0;
-						}
-					}
-				}
-			}
-			_pReslt = _evpTimer ^ _attack;
-			updateRegs();
-		}
-
-		int32 finOut = 0;
-		for (int ii = 0; ii < 3; ii++) {
-			int32 finOutTemp = ((_channels[ii].vol >> 4) & 1) ? _tleTable[_channels[ii].out ? _pReslt : 0] : _tlTable[_channels[ii].out ? (_channels[ii].vol & 0x0f) : 0];
-
-			if ((1 << ii) & _volMaskA)
-				finOutTemp = (finOutTemp * _volumeA) / Audio::Mixer::kMaxMixerVolume;
-
-			if ((1 << ii) & _volMaskB)
-				finOutTemp = (finOutTemp * _volumeB) / Audio::Mixer::kMaxMixerVolume;
-
-			finOut += finOutTemp;
-		}
-
-		finOut /= 3;
-
-		buffer[i << 1] += finOut;
-		buffer[(i << 1) + 1] += finOut;
-	}
-}
-
-void TownsPC98_FmSynthSquareSineSource::updateRegs() {
-	for (int i = 0; i < _updateRequest;) {
-		uint8 b = _updateRequestBuf[i++];
-		uint8 a = _updateRequestBuf[i++];
-		writeReg(a, b, true);
-	}
-	_updateRequest = -1;
-}
-
-#ifndef DISABLE_PC98_RHYTHM_CHANNEL
-TownsPC98_FmSynthPercussionSource::TownsPC98_FmSynthPercussionSource(const uint32 timerbase, const uint32 rtt) :
-	_rtt(rtt), _tickLength(timerbase * 2), _timer(0), _ready(false), _volMaskA(0), _volMaskB(0), _volumeA(Audio::Mixer::kMaxMixerVolume), _volumeB(Audio::Mixer::kMaxMixerVolume) {
-
-	memset(_rhChan, 0, sizeof(RhtChannel) * 6);
-	_reg = new uint8 *[40];
-
-	_reg[0] = _reg[1] = _reg[2] = _reg[3] = _reg[4] = _reg[5] = _reg[6] = _reg[7] = _reg[8] = _reg[9] = _reg[10] = _reg[11] = _reg[12] = _reg[13] = _reg[14] = _reg[15] = 0;
-	_reg[16] = &_rhChan[0].startPosL;
-	_reg[17] = &_rhChan[1].startPosL;
-	_reg[18] = &_rhChan[2].startPosL;
-	_reg[19] = &_rhChan[3].startPosL;
-	_reg[20] = &_rhChan[4].startPosL;
-	_reg[21] = &_rhChan[5].startPosL;
-	_reg[22] = &_rhChan[0].startPosH;
-	_reg[23] = &_rhChan[1].startPosH;
-	_reg[24] = &_rhChan[2].startPosH;
-	_reg[25] = &_rhChan[3].startPosH;
-	_reg[26] = &_rhChan[4].startPosH;
-	_reg[27] = &_rhChan[5].startPosH;
-	_reg[28] = &_rhChan[0].endPosL;
-	_reg[29] = &_rhChan[1].endPosL;
-	_reg[30] = &_rhChan[2].endPosL;
-	_reg[31] = &_rhChan[3].endPosL;
-	_reg[32] = &_rhChan[4].endPosL;
-	_reg[33] = &_rhChan[5].endPosL;
-	_reg[34] = &_rhChan[0].endPosH;
-	_reg[35] = &_rhChan[1].endPosH;
-	_reg[36] = &_rhChan[2].endPosH;
-	_reg[37] = &_rhChan[3].endPosH;
-	_reg[38] = &_rhChan[4].endPosH;
-	_reg[39] = &_rhChan[5].endPosH;
-}
-
-void TownsPC98_FmSynthPercussionSource::init(const uint8 *instrData) {
-	if (_ready) {
-		reset();
-		return;
-	}
-
-	const uint8 *start = instrData;
-	const uint8 *pos = start;
-
-	if (instrData) {
-		for (int i = 0; i < 6; i++) {
-			_rhChan[i].data = start + READ_BE_UINT16(pos);
-			pos += 2;
-			_rhChan[i].size = READ_BE_UINT16(pos);
-			pos += 2;
-		}
-		reset();
-		_ready = true;
-	} else {
-		memset(_rhChan, 0, sizeof(RhtChannel) * 6);
-		_ready = false;
-	}
-}
-
-void TownsPC98_FmSynthPercussionSource::reset() {
-	_timer = 0;
-	_totalLevel = 63;
-
-	for (int i = 0; i < 6; i++) {
-		RhtChannel *s = &_rhChan[i];
-		s->pos = s->start = s->data;
-		s->end = s->data + s->size;
-		s->active = false;
-		s->level = 0;
-		s->out = 0;
-		s->decStep = 1;
-		s->decState = 0;
-		s->samples[0] = s->samples[1] = 0;
-		s->startPosH = s->startPosL = s->endPosH = s->endPosL = 0;
-	}
-}
-
-void TownsPC98_FmSynthPercussionSource::writeReg(uint8 address, uint8 value) {
-	if (!_ready)
-		return;
-
-	uint8 h = address >> 4;
-	uint8 l = address & 15;
-
-	if (address > 15)
-		*_reg[address] = value;
-
-	if (address == 0) {
-		if (value & 0x80) {
-			//key off
-			for (int i = 0; i < 6; i++) {
-				if ((value >> i) & 1)
-					_rhChan[i].active = false;
-			}
-		} else {
-			//key on
-			for (int i = 0; i < 6; i++) {
-				if ((value >> i) & 1) {
-					RhtChannel *s = &_rhChan[i];
-					s->pos = s->start;
-					s->active = true;
-					s->out = 0;
-					s->samples[0] = s->samples[1] = 0;
-					s->decStep = 1;
-					s->decState = 0;
-				}
-			}
-		}
-	} else if (address == 1) {
-		// total level
-		_totalLevel = (value & 63) ^ 63;
-		for (int i = 0; i < 6; i++)
-			recalcOuput(&_rhChan[i]);
-	} else if (!h && l & 8) {
-		// instrument level
-		l &= 7;
-		_rhChan[l].level = (value & 0x1f) ^ 0x1f;
-		recalcOuput(&_rhChan[l]);
-	} else if (h & 3) {
-		l &= 7;
-		if (h == 1) {
-			// set start offset
-			_rhChan[l].start  = _rhChan[l].data + ((_rhChan[l].startPosH << 8 | _rhChan[l].startPosL) << 8);
-		} else if (h == 2) {
-			// set end offset
-			_rhChan[l].end = _rhChan[l].data + ((_rhChan[l].endPosH << 8 | _rhChan[l].endPosL) << 8) + 255;
-		}
-	}
-}
-
-void TownsPC98_FmSynthPercussionSource::nextTick(int32 *buffer, uint32 bufferSize) {
-	if (!_ready)
-		return;
-
-	for (uint32 i = 0; i < bufferSize; i++) {
-		_timer += _tickLength;
-		while (_timer > _rtt) {
-			_timer -= _rtt;
-
-			for (int ii = 0; ii < 6; ii++) {
-				RhtChannel *s = &_rhChan[ii];
-				if (s->active) {
-					recalcOuput(s);
-					if (s->decStep) {
-						advanceInput(s);
-						if (s->pos == s->end)
-							s->active = false;
-					}
-					s->decStep ^= 1;
-				}
-			}
-		}
-
-		int32 finOut = 0;
-
-		for (int ii = 0; ii < 6; ii++) {
-			if (_rhChan[ii].active)
-				finOut += _rhChan[ii].out;
-		}
-
-		finOut <<= 1;
-
-		if (1 & _volMaskA)
-			finOut = (finOut * _volumeA) / Audio::Mixer::kMaxMixerVolume;
-
-		if (1 & _volMaskB)
-			finOut = (finOut * _volumeB) / Audio::Mixer::kMaxMixerVolume;
-
-		buffer[i << 1] += finOut;
-		buffer[(i << 1) + 1] += finOut;
-	}
-}
-
-void TownsPC98_FmSynthPercussionSource::recalcOuput(RhtChannel *ins) {
-	uint32 s = _totalLevel + ins->level;
-	uint32 x = s > 62 ? 0 : (1 + (s >> 3));
-	int32 y = s > 62 ? 0 : (15 - (s & 7));
-	ins->out = ((ins->samples[ins->decStep] * y) >> x) & ~3;
-}
-
-void TownsPC98_FmSynthPercussionSource::advanceInput(RhtChannel *ins) {
-	static const int8 adjustIndex[] = { -1, -1, -1, -1, 2, 5, 7, 9 };
-
-	static const int16 stepTable[] = {
-		16, 17, 19, 21, 23, 25, 28, 31, 34, 37, 41, 45, 50, 55,
-		60, 66, 73, 80, 88, 97, 107, 118, 130, 143, 157, 173, 190, 209, 230, 253, 279, 307, 337,
-		371, 408, 449, 494, 544, 598, 658, 724, 796, 876, 963, 1060, 1166, 1282, 1411, 1552
-	};
-
-	uint8 cur = (int8)*ins->pos++;
-
-	for (int i = 0; i < 2; i++) {
-		int b = (2 * (cur & 7) + 1) * stepTable[ins->decState] / 8;
-		ins->samples[i] = CLIP<int16>(ins->samples[i ^ 1] + (cur & 8 ? b : -b), -2048, 2047);
-		ins->decState = CLIP<int8>(ins->decState + adjustIndex[cur & 7], 0, 48);
-		cur >>= 4;
-	}
-}
-#endif // DISABLE_PC98_RHYTHM_CHANNEL
-
-TownsPC98_FmSynth::TownsPC98_FmSynth(Audio::Mixer *mixer, EmuType type) :
-	_mixer(mixer),
-	_chanInternal(0), _ssg(0),
-#ifndef DISABLE_PC98_RHYTHM_CHANNEL
-	_prc(0),
-#endif
-	_numChan(type == kType26 ? 3 : 6), _numSSG(type == kTypeTowns ? 0 : 3),
-	_hasPercussion(type == kType86 ? true : false),
-	_oprRates(0), _oprRateshift(0), _oprAttackDecay(0), _oprFrq(0), _oprSinTbl(0), _oprLevelOut(0), _oprDetune(0),
-	 _rtt(type == kTypeTowns ? 0x514767 : 0x5B8D80), _baserate(55125.0f / (float)mixer->getOutputRate()),
-	_volMaskA(0), _volMaskB(0), _volumeA(255), _volumeB(255),
-	_regProtectionFlag(false), _ready(false) {
-
-	memset(&_timers[0], 0, sizeof(ChipTimer));
-	memset(&_timers[1], 0, sizeof(ChipTimer));
-
-	_timers[0].cb = _timers[1].cb = &TownsPC98_FmSynth::idleTimerCallback;
-	_timerbase = (uint32)(_baserate * 1000000.0f);
-}
-
-TownsPC98_FmSynth::~TownsPC98_FmSynth() {
-	if (_ready)
-		deinit();
-
-	delete _ssg;
-#ifndef DISABLE_PC98_RHYTHM_CHANNEL
-	delete _prc;
-#endif
-	delete[] _chanInternal;
-
-	delete[] _oprRates;
-	delete[] _oprRateshift;
-	delete[] _oprFrq;
-	delete[] _oprAttackDecay;
-	delete[] _oprSinTbl;
-	delete[] _oprLevelOut;
-	delete[] _oprDetune;
-}
-
-bool TownsPC98_FmSynth::init() {
-	if (_ready) {
-		reset();
-		return true;
-	}
-
-	generateTables();
-
-	_chanInternal = new ChanInternal[_numChan];
-	for (int i = 0; i < _numChan; i++) {
-		memset(&_chanInternal[i], 0, sizeof(ChanInternal));
-		for (int j = 0; j < 4; ++j)
-			_chanInternal[i].opr[j] = new TownsPC98_FmSynthOperator(_timerbase, _rtt, _oprRates, _oprRateshift, _oprAttackDecay, _oprFrq, _oprSinTbl, _oprLevelOut, _oprDetune);
-	}
-
-	if (_numSSG) {
-		_ssg = new TownsPC98_FmSynthSquareSineSource(_timerbase, _rtt);
-		_ssg->init(&_ssgTables[0], &_ssgTables[16]);
-	}
-
-#ifndef DISABLE_PC98_RHYTHM_CHANNEL
-	if (_hasPercussion) {
-		_prc = new TownsPC98_FmSynthPercussionSource(_timerbase, _rtt);
-		_prc->init(_percussionData);
-	}
-#endif
-
-	_timers[0].cb = &TownsPC98_FmSynth::timerCallbackA;
-	_timers[1].cb = &TownsPC98_FmSynth::timerCallbackB;
-
-	_mixer->playStream(Audio::Mixer::kPlainSoundType,
-	                   &_soundHandle, this, -1, Audio::Mixer::kMaxChannelVolume, 0, DisposeAfterUse::NO, true);
-
-	_ready = true;
-
-	return true;
-}
-
-void TownsPC98_FmSynth::reset() {
-	Common::StackLock lock(_mutex);
-	for (int i = 0; i < _numChan; i++) {
-		for (int ii = 0; ii < 4; ii++)
-			_chanInternal[i].opr[ii]->reset();
-		memset(_chanInternal[i].feedbuf, 0, 3);
-		_chanInternal[i].algorithm = 0;
-		_chanInternal[i].frqTemp = 0;
-		_chanInternal[i].enableLeft = _chanInternal[i].enableRight = true;
-		_chanInternal[i].updateEnvelopeParameters = false;
-	}
-
-	writeReg(0, 0x27, 0x33);
-
-	if (_ssg)
-		_ssg->reset();
-
-#ifndef DISABLE_PC98_RHYTHM_CHANNEL
-	if (_prc)
-		_prc->reset();
-#endif
-}
-
-void TownsPC98_FmSynth::writeReg(uint8 part, uint8 regAddress, uint8 value) {
-	if (_regProtectionFlag || !_ready)
-		return;
-
-	static const uint8 oprOrdr[] = { 0, 2, 1, 3 };
-
-	Common::StackLock lock(_mutex);
-
-	uint8 h = regAddress & 0xf0;
-	uint8 l = (regAddress & 0x0f);
-
-	ChanInternal *c = 0;
-	TownsPC98_FmSynthOperator **co = 0;
-	TownsPC98_FmSynthOperator *o = 0;
-
-	if (regAddress > 0x2F) {
-		c = &_chanInternal[(l & 3) + 3 * part];
-		co = c->opr;
-		o = c->opr[oprOrdr[(l - (l & 3)) >> 2]];
-	} else if (regAddress == 0x28) {
-		c = &_chanInternal[(value & 3) + ((value & 4) ? 3 : 0)];
-		co = c->opr;
-	}
-
-	switch (h) {
-	case 0x00:
-		// ssg
-		if (_ssg)
-			_ssg->writeReg(l, value);
-		break;
-	case 0x10:
-#ifndef DISABLE_PC98_RHYTHM_CHANNEL
-		// pcm rhythm channel
-		if (_prc)
-			_prc->writeReg(l, value);
-#endif
-		break;
-	case 0x20:
-		if (l == 8) {
-			// Key on/off
-			for (int i = 0; i < 4; i++) {
-				if ((value >> (4 + i)) & 1)
-					co[oprOrdr[i]]->keyOn();
-				else
-					co[oprOrdr[i]]->keyOff();
-			}
-		} else if (l == 4) {
-			// Timer A
-			_timers[0].value = (_timers[0].value & 3) | (value << 2);
-		} else if (l == 5) {
-			// Timer A
-			_timers[0].value = (_timers[0].value & 0x3fc) | (value & 3);
-		} else if (l == 6) {
-			// Timer B
-			_timers[1].value = value & 0xff;
-		} else if (l == 7) {
-			if (value & 1) {
-				float spc = (float)(0x400 - _timers[0].value) / _baserate;
-				if (spc < 1) {
-					warning("TownsPC98_FmSynth: Invalid Timer A setting: %d", _timers[0].value);
-					spc = 1;
-				}
-
-				_timers[0].smpPerCb = (int32) spc;
-				_timers[0].smpPerCbRem = (uint32)((spc - (float)_timers[0].smpPerCb) * 1000000.0f);
-				_timers[0].smpTillCb = _timers[0].smpPerCb;
-				_timers[0].smpTillCbRem = _timers[0].smpPerCbRem;
-				_timers[0].enabled = true;
-			} else {
-				_timers[0].enabled = false;
-			}
-
-			if (value & 2) {
-				float spc = (float)(0x100 - _timers[1].value) * 16.0f / _baserate;
-				if (spc < 1) {
-					warning("TownsPC98_FmSynth: Invalid Timer B setting: %d", _timers[1].value);
-					spc = 1;
-				}
-
-				_timers[1].smpPerCb = (int32) spc;
-				_timers[1].smpPerCbRem = (uint32)((spc - (float)_timers[1].smpPerCb) * 1000000.0f);
-				_timers[1].smpTillCb = _timers[1].smpPerCb;
-				_timers[1].smpTillCbRem = _timers[1].smpPerCbRem;
-				_timers[1].enabled = true;
-			} else {
-				_timers[1].enabled = false;
-			}
-
-			if (value & 0x10) {
-				_timers[0].smpTillCb = _timers[0].smpPerCb;
-				_timers[0].smpTillCbRem = _timers[0].smpTillCbRem;
-			}
-
-			if (value & 0x20) {
-				_timers[1].smpTillCb = _timers[1].smpPerCb;
-				_timers[1].smpTillCbRem = _timers[1].smpTillCbRem;
-			}
-		} else if (l == 2) {
-			// LFO
-			if (value & 8)
-				warning("TownsPC98_FmSynth: TRYING TO USE LFO (NOT SUPPORTED)");
-		} else if (l == 10 || l == 11) {
-			// DAC
-			if (l == 11 && (value & 0x80))
-				warning("TownsPC98_FmSynth: TRYING TO USE DAC (NOT SUPPORTED)");
-		}
-		break;
-
-	case 0x30:
-		// detune, multiple
-		o->detune((value >> 4) & 7);
-		o->multiple(value & 0x0f);
-		c->updateEnvelopeParameters = true;
-		break;
-
-	case 0x40:
-		// total level
-		o->totalLevel(value & 0x7f);
-		break;
-
-	case 0x50:
-		// rate scaling, attack rate
-		o->attackRate(value & 0x1f);
-		if (o->scaleRate(value >> 6))
-			c->updateEnvelopeParameters = true;
-		break;
-
-	case 0x60:
-		// first decay rate, amplitude modulation
-		o->decayRate(value & 0x1f);
-		o->ampModulation(value & 0x80 ? true : false);
-		break;
-
-	case 0x70:
-		// secondary decay rate
-		o->sustainRate(value & 0x1f);
-		break;
-
-	case 0x80:
-		// secondary amplitude, release rate;
-		o->sustainLevel(value >> 4);
-		o->releaseRate(value & 0x0f);
-		break;
-
-	case 0x90:
-		warning("TownsPC98_FmSynth: TRYING TO USE SSG ENVELOPE SHAPES (NOT SUPPORTED)");
-		break;
-
-	case 0xa0:
-		// frequency
-		l &= ~3;
-		if (l == 0) {
-			c->frqTemp = (c->frqTemp & 0xff00) | value;
-			c->updateEnvelopeParameters = true;
-			for (int i = 0; i < 4; i++)
-				co[i]->frequency(c->frqTemp);
-		} else if (l == 4) {
-			c->frqTemp = (c->frqTemp & 0xff) | (value << 8);
-		} else if (l == 8) {
-			// Ch 3/6 special mode frq
-			warning("TownsPC98_FmSynth: TRYING TO USE CH 3/6 SPECIAL MODE FREQ (NOT SUPPORTED)");
-		} else if (l == 12) {
-			// Ch 3/6 special mode frq
-			warning("TownsPC98_FmSynth: TRYING TO USE CH 3/6 SPECIAL MODE FREQ (NOT SUPPORTED)");
-		}
-		break;
-
-	case 0xb0:
-		l &= ~3;
-		if (l == 0) {
-			// feedback, _algorithm
-			co[0]->feedbackLevel((value >> 3) & 7);
-			c->algorithm = value & 7;
-		} else if (l == 4) {
-			// stereo, LFO sensitivity
-			c->enableLeft = value & 0x80 ? true : false;
-			c->enableRight = value & 0x40 ? true : false;
-			c->ampModSensitivity((value & 0x30) >> 4);
-			c->frqModSensitivity(value & 3);
-		}
-		break;
-
-	default:
-		warning("TownsPC98_FmSynth: UNKNOWN ADDRESS %d", regAddress);
-	}
-}
-
-int TownsPC98_FmSynth::readBuffer(int16 *buffer, const int numSamples) {
-	Common::StackLock lock(_mutex);
-
-	memset(buffer, 0, sizeof(int16) * numSamples);
-	int32 *tmp = new int32[numSamples];
-	int32 *tmpStart = tmp;
-	memset(tmp, 0, sizeof(int32) * numSamples);
-	int32 samplesLeft = numSamples >> 1;
-
-	while (_ready && samplesLeft) {
-		int32 render = samplesLeft;
-
-		for (int i = 0; i < 2; i++) {
-			if (_timers[i].enabled && _timers[i].cb) {
-				if (!_timers[i].smpTillCb) {
-					(this->*_timers[i].cb)();
-					_timers[i].smpTillCb = _timers[i].smpPerCb;
-
-					_timers[i].smpTillCbRem += _timers[i].smpPerCbRem;
-					if (_timers[i].smpTillCbRem >= _timerbase) {
-						_timers[i].smpTillCb++;
-						_timers[i].smpTillCbRem -= _timerbase;
-					}
-				}
-				render = MIN(render, _timers[i].smpTillCb);
-			}
-		}
-
-		samplesLeft -= render;
-
-		for (int i = 0; i < 2; i++) {
-			if (_timers[i].enabled && _timers[i].cb) {
-				_timers[i].smpTillCb -= render;
-			}
-		}
-
-		nextTick(tmp, render);
-
-		if (_ssg)
-			_ssg->nextTick(tmp, render);
-#ifndef DISABLE_PC98_RHYTHM_CHANNEL
-		if (_prc)
-			_prc->nextTick(tmp, render);
-#endif
-
-		nextTickEx(tmp, render);
-
-		for (int i = 0; i < render; ++i) {
-			int32 l = CLIP<int32>(tmp[i << 1], -32767, 32767);
-			buffer[i << 1] = (int16) l;
-			int32 r = CLIP<int32>(tmp[(i << 1) + 1], -32767, 32767);
-			buffer[(i << 1) + 1] = (int16) r;
-		}
-
-		buffer += (render << 1);
-		tmp += (render << 1);
-	}
-
-	delete[] tmpStart;
-	return numSamples;
-}
-
-void TownsPC98_FmSynth::deinit() {
-	_ready = false;
-	_mixer->stopHandle(_soundHandle);
-	_timers[0].cb = _timers[1].cb = &TownsPC98_FmSynth::idleTimerCallback;
-}
-
-uint8 TownsPC98_FmSynth::readSSGStatus() {
-	return _ssg->chanEnable();
-}
-
-void TownsPC98_FmSynth::setVolumeIntern(int volA, int volB) {
-	Common::StackLock lock(_mutex);
-	_volumeA = CLIP<uint16>(volA, 0, Audio::Mixer::kMaxMixerVolume);
-	_volumeB = CLIP<uint16>(volB, 0, Audio::Mixer::kMaxMixerVolume);
-	if (_ssg)
-		_ssg->setVolumeIntern(_volumeA, _volumeB);
-#ifndef DISABLE_PC98_RHYTHM_CHANNEL
-	if (_prc)
-		_prc->setVolumeIntern(_volumeA, _volumeB);
-#endif
-}
-
-void TownsPC98_FmSynth::setVolumeChannelMasks(int channelMaskA, int channelMaskB) {
-	Common::StackLock lock(_mutex);
-	_volMaskA = channelMaskA;
-	_volMaskB = channelMaskB;
-	if (_ssg)
-		_ssg->setVolumeChannelMasks(_volMaskA >> _numChan, _volMaskB >> _numChan);
-#ifndef DISABLE_PC98_RHYTHM_CHANNEL
-	if (_prc)
-		_prc->setVolumeChannelMasks(_volMaskA >> (_numChan + _numSSG), _volMaskB >> (_numChan + _numSSG));
-#endif
-}
-
-void TownsPC98_FmSynth::generateTables() {
-	delete[] _oprRates;
-	_oprRates = new uint8[128];
-
-	WRITE_BE_UINT32(_oprRates + 32, _numChan == 6 ? 0x90900000 : 0x00081018);
-	WRITE_BE_UINT32(_oprRates + 36, _numChan == 6 ? 0x00001010 : 0x00081018);
-	memset(_oprRates, 0x90, 32);
-	memset(_oprRates + 96, 0x80, 32);
-	uint8 *dst = (uint8 *)_oprRates + 40;
-	for (int i = 0; i < 40; i += 4)
-		WRITE_BE_UINT32(dst + i, 0x00081018);
-	for (int i = 0; i < 48; i += 4)
-		WRITE_BE_UINT32(dst + i, 0x00081018);
-	dst += 40;
-	for (uint8 i = 0; i < 16; i ++) {
-		uint8 v = (i < 12) ? i : 12;
-		*dst++ = ((4 + v) << 3);
-	}
-
-	delete[] _oprRateshift;
-	_oprRateshift = new uint8[128];
-	memset(_oprRateshift, 0, 128);
-	dst = (uint8 *)_oprRateshift + 32;
-	for (int i = 11; i; i--) {
-		memset(dst, i, 4);
-		dst += 4;
-	}
-
-	delete[] _oprFrq;
-	_oprFrq = new uint32[0x1000];
-	for (uint32 i = 0; i < 0x1000; i++)
-		_oprFrq[i] = (uint32)(_baserate * (float)(i << 11));
-
-	delete[] _oprAttackDecay;
-	_oprAttackDecay = new uint8[152];
-	memset(_oprAttackDecay, 0, 152);
-	for (int i = 0; i < 36; i++)
-		WRITE_BE_UINT32(_oprAttackDecay + (i << 2), _adtStat[i]);
-
-	delete[] _oprSinTbl;
-	_oprSinTbl = new uint32[1024];
-	for (int i = 0; i < 1024; i++) {
-		double val = sin((double)(((i << 1) + 1) * PI / 1024.0));
-		double d_dcb = log(1.0 / (double)ABS(val)) / log(2.0) * 256.0;
-		int32 i_dcb = (int32)(2.0 * d_dcb);
-		i_dcb = (i_dcb & 1) ? (i_dcb >> 1) + 1 : (i_dcb >> 1);
-		_oprSinTbl[i] = (i_dcb << 1) + (val >= 0.0 ? 0 : 1);
-	}
-
-	delete[] _oprLevelOut;
-	_oprLevelOut = new int32[0x1a00];
-	for (int i = 0; i < 256; i++) {
-		double val = floor(65536.0 / pow(2.0, 0.00390625 * (double)(1 + i)));
-		int32 val_int = ((int32) val) >> 4;
-		_oprLevelOut[i << 1] = (val_int & 1) ? ((val_int >> 1) + 1) << 2 : (val_int >> 1) << 2;
-		_oprLevelOut[(i << 1) + 1] = -_oprLevelOut[i << 1];
-		for (int ii = 1; ii < 13; ++ii) {
-			_oprLevelOut[(i << 1) + (ii << 9)] =  _oprLevelOut[i << 1] >> ii;
-			_oprLevelOut[(i << 1) + (ii << 9) + 1] = -_oprLevelOut[(i << 1) + (ii << 9)];
-		}
-	}
-
-	uint8 *dtt = new uint8[128];
-	memset(dtt, 0, 36);
-	memset(dtt + 36, 1, 8);
-	memcpy(dtt + 44, _detSrc, 84);
-
-	delete[] _oprDetune;
-	_oprDetune = new int32[256];
-	for (int i = 0; i < 128; i++) {
-		_oprDetune[i] = (int32)((float)dtt[i] * _baserate * 64.0);
-		_oprDetune[i + 128] = -_oprDetune[i];
-	}
-
-	delete[] dtt;
-}
-
-void TownsPC98_FmSynth::nextTick(int32 *buffer, uint32 bufferSize) {
-	if (!_ready)
-		return;
-
-	for (int i = 0; i < _numChan; i++) {
-		TownsPC98_FmSynthOperator **o = _chanInternal[i].opr;
-
-		if (_chanInternal[i].updateEnvelopeParameters) {
-			_chanInternal[i].updateEnvelopeParameters = false;
-			for (int ii = 0; ii < 4 ; ii++)
-				o[ii]->updatePhaseIncrement();
-		}
-
-		for (uint32 ii = 0; ii < bufferSize ; ii++) {
-			int32 phbuf1, phbuf2, output;
-			phbuf1 = phbuf2 = output = 0;
-
-			int32 *leftSample = &buffer[ii * 2];
-			int32 *rightSample = &buffer[ii * 2 + 1];
-			int32 *del = &_chanInternal[i].feedbuf[2];
-			int32 *feed = _chanInternal[i].feedbuf;
-
-			switch (_chanInternal[i].algorithm) {
-			case 0:
-				o[0]->generateOutput(0, feed, phbuf1);
-				o[2]->generateOutput(*del, 0, phbuf2);
-				*del = 0;
-				o[1]->generateOutput(phbuf1, 0, *del);
-				o[3]->generateOutput(phbuf2, 0, output);
-				break;
-			case 1:
-				o[0]->generateOutput(0, feed, phbuf1);
-				o[2]->generateOutput(*del, 0, phbuf2);
-				o[1]->generateOutput(0, 0, phbuf1);
-				o[3]->generateOutput(phbuf2, 0, output);
-				*del = phbuf1;
-				break;
-			case 2:
-				o[0]->generateOutput(0, feed, phbuf2);
-				o[2]->generateOutput(*del, 0, phbuf2);
-				o[1]->generateOutput(0, 0, phbuf1);
-				o[3]->generateOutput(phbuf2, 0, output);
-				*del = phbuf1;
-				break;
-			case 3:
-				o[0]->generateOutput(0, feed, phbuf2);
-				o[2]->generateOutput(0, 0, *del);
-				o[1]->generateOutput(phbuf2, 0, phbuf1);
-				o[3]->generateOutput(*del, 0, output);
-				*del = phbuf1;
-				break;
-			case 4:
-				o[0]->generateOutput(0, feed, phbuf1);
-				o[2]->generateOutput(0, 0, phbuf2);
-				o[1]->generateOutput(phbuf1, 0, output);
-				o[3]->generateOutput(phbuf2, 0, output);
-				*del = 0;
-				break;
-			case 5:
-				o[0]->generateOutput(0, feed, phbuf1);
-				o[2]->generateOutput(*del, 0, output);
-				o[1]->generateOutput(phbuf1, 0, output);
-				o[3]->generateOutput(phbuf1, 0, output);
-				*del = phbuf1;
-				break;
-			case 6:
-				o[0]->generateOutput(0, feed, phbuf1);
-				o[2]->generateOutput(0, 0, output);
-				o[1]->generateOutput(phbuf1, 0, output);
-				o[3]->generateOutput(0, 0, output);
-				*del = 0;
-				break;
-			case 7:
-				o[0]->generateOutput(0, feed, output);
-				o[2]->generateOutput(0, 0, output);
-				o[1]->generateOutput(0, 0, output);
-				o[3]->generateOutput(0, 0, output);
-				*del = 0;
-				break;
-			};
-
-			int32 finOut = (output << 2) / ((_numChan + _numSSG - 3) / 3);
-
-			if ((1 << i) & _volMaskA)
-				finOut = (finOut * _volumeA) / Audio::Mixer::kMaxMixerVolume;
-
-			if ((1 << i) & _volMaskB)
-				finOut = (finOut * _volumeB) / Audio::Mixer::kMaxMixerVolume;
-
-			if (_chanInternal[i].enableLeft)
-				*leftSample += finOut;
-
-			if (_chanInternal[i].enableRight)
-				*rightSample += finOut;
-		}
-	}
-}
-
-const uint32 TownsPC98_FmSynth::_adtStat[] = {
-	0x00010001, 0x00010001, 0x00010001, 0x01010001,
-	0x00010101, 0x00010101, 0x00010101, 0x01010101,
-	0x01010101, 0x01010101, 0x01010102, 0x01010102,
-	0x01020102, 0x01020102, 0x01020202, 0x01020202,
-	0x02020202, 0x02020202, 0x02020204, 0x02020204,
-	0x02040204, 0x02040204, 0x02040404, 0x02040404,
-	0x04040404, 0x04040404, 0x04040408, 0x04040408,
-	0x04080408, 0x04080408, 0x04080808, 0x04080808,
-	0x08080808, 0x08080808, 0x10101010, 0x10101010
-};
-
-const uint8 TownsPC98_FmSynth::_detSrc[] = {
-	0x02, 0x02, 0x02, 0x02, 0x02, 0x03, 0x03, 0x03,
-	0x04, 0x04, 0x04, 0x05, 0x05, 0x06, 0x06, 0x07,
-	0x08, 0x08, 0x08, 0x08, 0x01, 0x01, 0x01, 0x01,
-	0x02, 0x02, 0x02, 0x02, 0x02, 0x03, 0x03, 0x03,
-	0x04, 0x04, 0x04, 0x05, 0x05, 0x06, 0x06, 0x07,
-	0x08, 0x08, 0x09, 0x0a, 0x0b, 0x0c, 0x0d, 0x0e,
-	0x10, 0x10, 0x10, 0x10, 0x02, 0x02, 0x02, 0x02,
-	0x02, 0x03, 0x03, 0x03, 0x04, 0x04, 0x04, 0x05,
-	0x05, 0x06, 0x06, 0x07, 0x08, 0x08, 0x09, 0x0a,
-	0x0b, 0x0c, 0x0d, 0x0e, 0x10, 0x11, 0x13, 0x14,
-	0x16, 0x16, 0x16, 0x16
-};
-
-const int TownsPC98_FmSynth::_ssgTables[] = {
-	0x01202A, 0x0092D2, 0x006B42, 0x0053CB, 0x003DF8, 0x003053, 0x0022DA, 0x001A8C,
-	0x00129B, 0x000DC1, 0x000963, 0x0006C9, 0x000463, 0x0002FA, 0x0001B6, 0x0000FB,
-	0x0193B6, 0x01202A, 0x00CDB1, 0x0092D2, 0x007D7D, 0x006B42, 0x005ECD, 0x0053CB,
-	0x00480F, 0x003DF8, 0x0036B9, 0x003053, 0x00290A, 0x0022DA, 0x001E6B, 0x001A8C,
-	0x001639, 0x00129B, 0x000FFF, 0x000DC1, 0x000B5D, 0x000963, 0x0007FB, 0x0006C9,
-	0x000575, 0x000463, 0x00039D, 0x0002FA, 0x000242, 0x0001B6, 0x00014C, 0x0000FB
-};
-
-#ifndef DISABLE_PC98_RHYTHM_CHANNEL
-const uint8 TownsPC98_FmSynth::_percussionData[] = {
-	0, 24, 1, 192, 1, 216, 2, 128, 4, 88, 23, 64, 27, 152, 1, 128, 29, 24, 2, 128, 31, 152, 0, 128, 136, 128, 128, 128, 0, 136, 97, 103, 153, 139, 34, 163, 72, 195, 27, 69, 1, 154, 137, 35, 8, 51, 169, 122, 164, 75, 133, 203, 81, 146, 168, 121, 185, 68, 202, 8, 33, 237, 49, 177, 12, 133, 140, 17, 160, 42, 161, 10, 0, 137, 176, 57,
-	233, 41, 160, 136, 235, 65, 177, 137, 128, 26, 164, 28, 3, 157, 51, 137, 1, 152, 113, 161, 40, 146, 115, 192, 56, 5, 169, 66, 161, 56, 1, 50, 145, 59, 39, 168, 97, 1, 160, 57, 7, 153, 50, 153, 32, 2, 25, 129, 32, 20, 186, 66, 129, 24, 153, 164, 142, 130, 169, 153, 26, 242, 138, 217, 9, 128, 204, 58, 209, 172, 40, 176, 141,
-	128, 155, 144, 203, 139, 0, 235, 9, 177, 172, 0, 185, 168, 138, 25, 240, 59, 211, 139, 19, 176, 90, 160, 17, 26, 132, 41, 1, 5, 25, 3, 50, 144, 115, 147, 42, 39, 152, 41, 3, 56, 193, 105, 130, 155, 66, 200, 26, 19, 218, 154, 49, 201, 171, 138, 176, 251, 139, 185, 172, 136, 189, 139, 145, 207, 41, 160, 171, 152, 186, 139,
-	186, 141, 128, 218, 171, 51, 217, 170, 56, 163, 12, 4, 155, 81, 147, 42, 37, 152, 32, 54, 136, 49, 50, 48, 37, 32, 69, 0, 17, 50, 50, 83, 2, 16, 68, 20, 8, 66, 4, 154, 84, 145, 24, 33, 24, 32, 17, 18, 145, 32, 22, 168, 49, 163, 1, 33, 50, 184, 115, 129, 25, 66, 1, 24, 67, 2, 80, 35, 40, 53, 2, 65, 51, 19, 67, 37, 0, 52, 35, 49, 37,
-	34, 49, 37, 17, 52, 17, 35, 35, 35, 34, 32, 49, 33, 152, 34, 145, 24, 24, 128, 138, 128, 184, 9, 177, 171, 168, 185, 155, 152, 172, 155, 186, 172, 185, 172, 155, 186, 173, 153, 202, 187, 185, 202, 170, 171, 202, 186, 169, 170, 170, 171, 139, 154, 171, 153, 154, 169, 10, 168, 154, 128, 168, 154, 0, 153, 152, 136, 137,
-	128, 153, 0, 152, 8, 128, 137, 0, 136, 136, 8, 9, 8, 9, 8, 24, 153, 128, 136, 153, 144, 0, 161, 138, 1, 169, 136, 128, 160, 168, 152, 153, 138, 137, 154, 153, 153, 154, 153, 170, 168, 170, 185, 168, 169, 154, 169, 171, 153, 169, 170, 153, 152, 154, 153, 137, 169, 137, 136, 144, 152, 144, 128, 128, 144, 129, 129, 0, 33,
-	0, 17, 17, 17, 33, 33, 18, 18, 34, 34, 34, 34, 34, 34, 35, 19, 35, 19, 35, 35, 18, 19, 18, 35, 18, 33, 0, 8, 8, 8, 8, 8, 8, 8, 160, 205, 65, 176, 171, 203, 16, 240, 95, 242, 120, 145, 156, 66, 177, 26, 19, 153, 9, 35, 35, 239, 56, 132, 138, 154, 50, 145, 203, 25, 32, 20, 237, 24, 130, 138, 160, 27, 39, 173, 50, 203, 64, 145, 139,
-	18, 168, 48, 146, 171, 65, 18, 176, 12, 52, 128, 25, 5, 57, 240, 104, 161, 25, 129, 18, 188, 114, 160, 26, 36, 200, 154, 18, 1, 128, 186, 73, 162, 173, 32, 184, 25, 144, 137, 234, 8, 154, 32, 160, 158, 18, 187, 81, 2, 235, 41, 36, 144, 154, 17, 67, 128, 33, 160, 114, 146, 26, 37, 33, 232, 41, 130, 41, 178, 29, 50, 251, 24,
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-	33, 52, 53, 83, 50, 51, 52, 52, 37, 34, 34, 18, 16, 144, 152, 154, 187, 219, 203, 188, 173, 186, 186, 186, 170, 154, 153, 138, 144, 16, 17, 67, 82, 50, 51, 21, 34, 19, 33, 2, 18, 33, 1, 8, 153, 169, 153, 153, 136, 128, 0, 136, 154, 153, 153, 8, 8, 1, 16, 0, 169, 170, 187, 171, 171, 154, 153, 153, 152, 153, 153, 0, 16, 51, 83,
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-	45, 136, 18, 144, 105, 138, 1, 160, 14, 128, 132, 145, 186, 37, 138, 41, 192, 48, 145, 46, 160, 33, 44, 24, 225, 16, 13, 132, 136, 137, 16, 148, 25, 170, 194, 82, 152, 136, 91, 24, 42, 169, 33, 233, 131, 179, 24, 185, 149, 16, 57, 172, 164, 18, 10, 211, 160, 147, 211, 33, 138, 243, 129, 16, 41, 193, 0, 43, 132, 155, 73,
-	58, 145, 244, 145, 43, 35, 9, 171, 16, 110, 25, 8, 28, 74, 162, 128, 26, 27, 82, 45, 136, 153, 18, 8, 136, 8
-};
-#endif // DISABLE_PC98_RHYTHM_CHANNEL
-
-TownsPC98_FmSynth::ChanInternal::ChanInternal() {
-	memset(this, 0, sizeof(ChanInternal));
-}
-
-TownsPC98_FmSynth::ChanInternal::~ChanInternal() {
-	for (uint i = 0; i < ARRAYSIZE(opr); ++i)
-		delete opr[i];
-}
diff --git a/sound/softsynth/fmtowns_pc98/towns_pc98_fmsynth.h b/sound/softsynth/fmtowns_pc98/towns_pc98_fmsynth.h
deleted file mode 100644
index ddd249b1b8..0000000000
--- a/sound/softsynth/fmtowns_pc98/towns_pc98_fmsynth.h
+++ /dev/null
@@ -1,196 +0,0 @@
-/* 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 TOWNS_PC98_FMSYNTH_H
-#define TOWNS_PC98_FMSYNTH_H
-
-#include "sound/audiostream.h"
-#include "sound/mixer.h"
-#include "common/list.h"
-
-#ifdef __DS__
-/* This disables the rhythm channel when emulating the PC-98 type 86 sound card.
- * The only purpose is code size reduction for certain backends.
- * At the moment the only games which make use of the rhythm channel are the
- * (very rare) PC-98 versions of Legend of Kyrandia 2 and Lands of Lore. Music will
- * still be okay, just missing a couple of rhythm instruments.
- */
-#define DISABLE_PC98_RHYTHM_CHANNEL
-#endif
-
-class TownsPC98_FmSynthOperator;
-class TownsPC98_FmSynthSquareSineSource;
-#ifndef DISABLE_PC98_RHYTHM_CHANNEL
-class TownsPC98_FmSynthPercussionSource;
-#endif
-
-enum EnvelopeState {
-	kEnvReady,
-	kEnvAttacking,
-	kEnvDecaying,
-	kEnvSustaining,
-	kEnvReleasing
-};
-
-class TownsPC98_FmSynth : public Audio::AudioStream {
-public:
-	enum EmuType {
-		kTypeTowns,
-		kType26,
-		kType86
-	};
-
-	TownsPC98_FmSynth(Audio::Mixer *mixer, EmuType type);
-	virtual ~TownsPC98_FmSynth();
-
-	virtual bool init();
-	virtual void reset();
-
-	void writeReg(uint8 part, uint8 regAddress, uint8 value);
-
-	// AudioStream interface
-	int readBuffer(int16 *buffer, const int numSamples);
-	bool isStereo() const {
-		return true;
-	}
-	bool endOfData() const {
-		return false;
-	}
-	int getRate() const {
-		return _mixer->getOutputRate();
-	}
-
-protected:
-	void deinit();
-
-	// Implement this in your inherited class if your driver generates
-	// additional output that has to be inserted into the buffer.
-	virtual void nextTickEx(int32 *buffer, uint32 bufferSize) {}
-
-	void toggleRegProtection(bool prot) {
-		_regProtectionFlag = prot;
-	}
-	uint8 readSSGStatus();
-
-	virtual void timerCallbackA() = 0;
-	virtual void timerCallbackB() = 0;
-
-	// The audio driver can store and apply two different audio settings
-	// (usually for music and sound effects). The channel mask will determine
-	// which channels get effected by the setting. The first bits will be
-	// the normal fm channels, the next bits the ssg channels and the final
-	// bit the rhythm channel.
-	void setVolumeIntern(int volA, int volB);
-	void setVolumeChannelMasks(int channelMaskA, int channelMaskB);
-
-	const int _numChan;
-	const int _numSSG;
-	const bool _hasPercussion;
-
-	Common::Mutex _mutex;
-private:
-	void generateTables();
-	void nextTick(int32 *buffer, uint32 bufferSize);
-	void generateOutput(int32 &leftSample, int32 &rightSample, int32 *del, int32 *feed);
-
-	struct ChanInternal {
-		ChanInternal();
-		~ChanInternal();
-
-		void ampModSensitivity(uint32 value) {
-			ampModSvty = (1 << (3 - value)) - (((value >> 1) & 1) | (value & 1));
-		}
-		void frqModSensitivity(uint32 value) {
-			frqModSvty = value << 5;
-		}
-
-		uint16 frqTemp;
-		bool enableLeft;
-		bool enableRight;
-		bool updateEnvelopeParameters;
-		int32 feedbuf[3];
-		uint8 algorithm;
-
-		uint32 ampModSvty;
-		uint32 frqModSvty;
-
-		TownsPC98_FmSynthOperator *opr[4];
-	};
-
-	TownsPC98_FmSynthSquareSineSource *_ssg;
-#ifndef DISABLE_PC98_RHYTHM_CHANNEL
-	TownsPC98_FmSynthPercussionSource *_prc;
-#endif
-	ChanInternal *_chanInternal;
-
-	uint8 *_oprRates;
-	uint8 *_oprRateshift;
-	uint8 *_oprAttackDecay;
-	uint32 *_oprFrq;
-	uint32 *_oprSinTbl;
-	int32 *_oprLevelOut;
-	int32 *_oprDetune;
-
-	bool _regProtectionFlag;
-
-	typedef void (TownsPC98_FmSynth::*ChipTimerProc)();
-	void idleTimerCallback() {}
-
-	struct ChipTimer {
-		bool enabled;
-		uint16 value;
-
-		int32 smpTillCb;
-		uint32 smpTillCbRem;
-		int32 smpPerCb;
-		uint32 smpPerCbRem;
-
-		ChipTimerProc cb;
-	};
-
-	ChipTimer _timers[2];
-
-	int _volMaskA, _volMaskB;
-	uint16 _volumeA, _volumeB;
-
-	const float _baserate;
-	uint32 _timerbase;
-	uint32 _rtt;
-
-	Audio::Mixer *_mixer;
-	Audio::SoundHandle _soundHandle;
-
-#ifndef DISABLE_PC98_RHYTHM_CHANNEL
-	static const uint8 _percussionData[];
-#endif
-	static const uint32 _adtStat[];
-	static const uint8 _detSrc[];
-	static const int _ssgTables[];
-
-	bool _ready;
-};
-
-#endif
-
diff --git a/sound/softsynth/mt32.cpp b/sound/softsynth/mt32.cpp
deleted file mode 100644
index 09e7e48151..0000000000
--- a/sound/softsynth/mt32.cpp
+++ /dev/null
@@ -1,573 +0,0 @@
-/* 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"
-
-#ifdef USE_MT32EMU
-
-#include "sound/softsynth/mt32/mt32emu.h"
-
-#include "sound/softsynth/emumidi.h"
-#include "sound/musicplugin.h"
-#include "sound/mpu401.h"
-
-#include "common/config-manager.h"
-#include "common/debug.h"
-#include "common/events.h"
-#include "common/file.h"
-#include "common/system.h"
-#include "common/util.h"
-#include "common/archive.h"
-#include "common/translation.h"
-
-#include "graphics/fontman.h"
-#include "graphics/surface.h"
-
-class MidiChannel_MT32 : public MidiChannel_MPU401 {
-	void effectLevel(byte value) { }
-	void chorusLevel(byte value) { }
-};
-
-class MidiDriver_MT32 : public MidiDriver_Emulated {
-private:
-	Audio::SoundHandle _handle;
-	MidiChannel_MT32 _midiChannels[16];
-	uint16 _channelMask;
-	MT32Emu::Synth *_synth;
-
-	int _outputRate;
-
-protected:
-	void generateSamples(int16 *buf, int len);
-
-public:
-	bool _initialising;
-
-	MidiDriver_MT32(Audio::Mixer *mixer);
-	virtual ~MidiDriver_MT32();
-
-	int open();
-	void close();
-	void send(uint32 b);
-	void setPitchBendRange (byte channel, uint range);
-	void sysEx(const byte *msg, uint16 length);
-
-	uint32 property(int prop, uint32 param);
-	MidiChannel *allocateChannel();
-	MidiChannel *getPercussionChannel();
-
-	// AudioStream API
-	bool isStereo() const { return true; }
-	int getRate() const { return _outputRate; }
-};
-
-class MT32File : public MT32Emu::File {
-	Common::File _in;
-	Common::DumpFile _out;
-public:
-	bool open(const char *filename, OpenMode mode) {
-		if (mode == OpenMode_read)
-			return _in.open(filename);
-		else
-			return _out.open(filename);
-	}
-	void close() {
-		_in.close();
-		_out.close();
-	}
-	size_t read(void *in, size_t size) {
-		return _in.read(in, size);
-	}
-	bool readBit8u(MT32Emu::Bit8u *in) {
-		byte b = _in.readByte();
-		if (_in.eos())
-			return false;
-		*in = b;
-		return true;
-	}
-	size_t write(const void *in, size_t size) {
-		return _out.write(in, size);
-	}
-	bool writeBit8u(MT32Emu::Bit8u out) {
-		_out.writeByte(out);
-		return !_out.err();
-	}
-	bool isEOF() {
-		return _in.isOpen() && _in.eos();
-	}
-};
-
-static int eatSystemEvents() {
-	Common::Event event;
-	Common::EventManager *eventMan = g_system->getEventManager();
-	while (eventMan->pollEvent(event)) {
-		switch (event.type) {
-		case Common::EVENT_QUIT:
-			return 1;
-		default:
-			break;
-		}
-	}
-	return 0;
-}
-
-static void drawProgress(float progress) {
-	const Graphics::Font &font(*FontMan.getFontByUsage(Graphics::FontManager::kOSDFont));
-	Graphics::Surface *screen = g_system->lockScreen();
-
-	assert(screen);
-	assert(screen->pixels);
-
-	Graphics::PixelFormat screenFormat = g_system->getScreenFormat();
-
-	int16 w = g_system->getWidth() / 7 * 5;
-	int16 h = font.getFontHeight();
-	int16 x = g_system->getWidth() / 7;
-	int16 y = g_system->getHeight() / 2 - h / 2;
-
-	Common::Rect r(x, y, x + w, y + h);
-
-	uint32 col;
-
-	if (screenFormat.bytesPerPixel > 1)
-		col = screenFormat.RGBToColor(0, 171, 0);
-	else
-		col = 1;
-
-	screen->frameRect(r, col);
-
-	r.grow(-1);
-	r.setWidth(uint16(progress * w));
-
-	if (screenFormat.bytesPerPixel > 1)
-		col = screenFormat.RGBToColor(171, 0, 0);
-	else
-		col = 2;
-
-	screen->fillRect(r, col);
-
-	g_system->unlockScreen();
-	g_system->updateScreen();
-}
-
-static void drawMessage(int offset, const Common::String &text) {
-	const Graphics::Font &font(*FontMan.getFontByUsage(Graphics::FontManager::kOSDFont));
-	Graphics::Surface *screen = g_system->lockScreen();
-
-	assert(screen);
-	assert(screen->pixels);
-
-	Graphics::PixelFormat screenFormat = g_system->getScreenFormat();
-
-	uint16 h = font.getFontHeight();
-	uint16 y = g_system->getHeight() / 2 - h / 2 + offset * (h + 1);
-
-	uint32 col;
-
-	if (screenFormat.bytesPerPixel > 1)
-		col = screenFormat.RGBToColor(0, 0, 0);
-	else
-		col = 0;
-
-	Common::Rect r(0, y, screen->w, y + h);
-	screen->fillRect(r, col);
-
-	if (screenFormat.bytesPerPixel > 1)
-		col = screenFormat.RGBToColor(0, 171, 0);
-	else
-		col = 1;
-
-	font.drawString(screen, text, 0, y, screen->w, col, Graphics::kTextAlignCenter);
-
-	g_system->unlockScreen();
-	g_system->updateScreen();
-}
-
-static MT32Emu::File *MT32_OpenFile(void *userData, const char *filename, MT32Emu::File::OpenMode mode) {
-	MT32File *file = new MT32File();
-	if (!file->open(filename, mode)) {
-		delete file;
-		return NULL;
-	}
-	return file;
-}
-
-static void MT32_PrintDebug(void *userData, const char *fmt, va_list list) {
-	if (((MidiDriver_MT32 *)userData)->_initialising) {
-		char buf[512];
-
-		vsnprintf(buf, 512, fmt, list);
-		buf[70] = 0; // Truncate to a reasonable length
-
-		drawMessage(1, buf);
-	}
-
-	//vdebug(0, fmt, list); // FIXME: Use a higher debug level
-}
-
-static int MT32_Report(void *userData, MT32Emu::ReportType type, const void *reportData) {
-	switch (type) {
-	case MT32Emu::ReportType_lcdMessage:
-		g_system->displayMessageOnOSD((const char *)reportData);
-		break;
-	case MT32Emu::ReportType_errorControlROM:
-		error("Failed to load MT32_CONTROL.ROM");
-		break;
-	case MT32Emu::ReportType_errorPCMROM:
-		error("Failed to load MT32_PCM.ROM");
-		break;
-	case MT32Emu::ReportType_progressInit:
-		if (((MidiDriver_MT32 *)userData)->_initialising) {
-			drawProgress(*((const float *)reportData));
-			return eatSystemEvents();
-		}
-		break;
-	case MT32Emu::ReportType_availableSSE:
-		debug(1, "MT32emu: SSE is available");
-		break;
-	case MT32Emu::ReportType_usingSSE:
-		debug(1, "MT32emu: using SSE");
-		break;
-	case MT32Emu::ReportType_available3DNow:
-		debug(1, "MT32emu: 3DNow! is available");
-		break;
-	case MT32Emu::ReportType_using3DNow:
-		debug(1, "MT32emu: using 3DNow!");
-		break;
-	default:
-		break;
-	}
-	return 0;
-}
-
-////////////////////////////////////////
-//
-// MidiDriver_MT32
-//
-////////////////////////////////////////
-
-MidiDriver_MT32::MidiDriver_MT32(Audio::Mixer *mixer) : MidiDriver_Emulated(mixer) {
-	_channelMask = 0xFFFF; // Permit all 16 channels by default
-	uint i;
-	for (i = 0; i < ARRAYSIZE(_midiChannels); ++i) {
-		_midiChannels[i].init(this, i);
-	}
-	_synth = NULL;
-	// A higher baseFreq reduces the length used in generateSamples(),
-	// and means that the timer callback will be called more often.
-	// That results in more accurate timing.
-	_baseFreq = 10000;
-	// Unfortunately bugs in the emulator cause inaccurate tuning
-	// at rates other than 32KHz, thus we produce data at 32KHz and
-	// rely on Mixer to convert.
-	_outputRate = 32000; //_mixer->getOutputRate();
-	_initialising = false;
-}
-
-MidiDriver_MT32::~MidiDriver_MT32() {
-	delete _synth;
-}
-
-int MidiDriver_MT32::open() {
-	MT32Emu::SynthProperties prop;
-
-	if (_isOpen)
-		return MERR_ALREADY_OPEN;
-
-	MidiDriver_Emulated::open();
-
-	memset(&prop, 0, sizeof(prop));
-	prop.sampleRate = getRate();
-	prop.useReverb = true;
-	prop.useDefaultReverb = false;
-	prop.reverbType = 0;
-	prop.reverbTime = 5;
-	prop.reverbLevel = 3;
-	prop.userData = this;
-	prop.printDebug = MT32_PrintDebug;
-	prop.report = MT32_Report;
-	prop.openFile = MT32_OpenFile;
-
-	_synth = new MT32Emu::Synth();
-
-	Graphics::PixelFormat screenFormat = g_system->getScreenFormat();
-
-	if (screenFormat.bytesPerPixel == 1) {
-		const byte dummy_palette[] = {
-			0, 0, 0, 0,		// background
-			0, 171, 0, 0,	// border, font
-			171, 0, 0, 0	// fill
-		};
-
-		g_system->getPaletteManager()->setPalette(dummy_palette, 0, 3);
-	}
-
-	_initialising = true;
-	drawMessage(-1, _s("Initialising MT-32 Emulator"));
-	if (!_synth->open(prop))
-		return MERR_DEVICE_NOT_AVAILABLE;
-	_initialising = false;
-
-	if (screenFormat.bytesPerPixel > 1)
-		g_system->fillScreen(screenFormat.RGBToColor(0, 0, 0));
-	else
-		g_system->fillScreen(0);
-
-	g_system->updateScreen();
-
-	_mixer->playStream(Audio::Mixer::kSFXSoundType, &_handle, this, -1, Audio::Mixer::kMaxChannelVolume, 0, DisposeAfterUse::NO, true);
-
-	return 0;
-}
-
-void MidiDriver_MT32::send(uint32 b) {
-	_synth->playMsg(b);
-}
-
-void MidiDriver_MT32::setPitchBendRange(byte channel, uint range) {
-	if (range > 24) {
-		printf("setPitchBendRange() called with range > 24: %d", range);
-	}
-	byte benderRangeSysex[9];
-	benderRangeSysex[0] = 0x41; // Roland
-	benderRangeSysex[1] = channel;
-	benderRangeSysex[2] = 0x16; // MT-32
-	benderRangeSysex[3] = 0x12; // Write
-	benderRangeSysex[4] = 0x00;
-	benderRangeSysex[5] = 0x00;
-	benderRangeSysex[6] = 0x04;
-	benderRangeSysex[7] = (byte)range;
-	benderRangeSysex[8] = MT32Emu::Synth::calcSysexChecksum(&benderRangeSysex[4], 4, 0);
-	sysEx(benderRangeSysex, 9);
-}
-
-void MidiDriver_MT32::sysEx(const byte *msg, uint16 length) {
-	if (msg[0] == 0xf0) {
-		_synth->playSysex(msg, length);
-	} else {
-		_synth->playSysexWithoutFraming(msg, length);
-	}
-}
-
-void MidiDriver_MT32::close() {
-	if (!_isOpen)
-		return;
-	_isOpen = false;
-
-	// Detach the player callback handler
-	setTimerCallback(NULL, NULL);
-	// Detach the mixer callback handler
-	_mixer->stopHandle(_handle);
-
-	_synth->close();
-	delete _synth;
-	_synth = NULL;
-}
-
-void MidiDriver_MT32::generateSamples(int16 *data, int len) {
-	_synth->render(data, len);
-}
-
-uint32 MidiDriver_MT32::property(int prop, uint32 param) {
-	switch (prop) {
-	case PROP_CHANNEL_MASK:
-		_channelMask = param & 0xFFFF;
-		return 1;
-	}
-
-	return 0;
-}
-
-MidiChannel *MidiDriver_MT32::allocateChannel() {
-	MidiChannel_MT32 *chan;
-	uint i;
-
-	for (i = 0; i < ARRAYSIZE(_midiChannels); ++i) {
-		if (i == 9 || !(_channelMask & (1 << i)))
-			continue;
-		chan = &_midiChannels[i];
-		if (chan->allocate()) {
-			return chan;
-		}
-	}
-	return NULL;
-}
-
-MidiChannel *MidiDriver_MT32::getPercussionChannel() {
-	return &_midiChannels[9];
-}
-
-// This code should be used when calling the timer callback from the mixer thread is undesirable.
-// Note that it results in less accurate timing.
-#if 0
-class MidiEvent_MT32 {
-public:
-	MidiEvent_MT32 *_next;
-	uint32 _msg; // 0xFFFFFFFF indicates a sysex message
-	byte *_data;
-	uint32 _len;
-
-	MidiEvent_MT32(uint32 msg, byte *data, uint32 len) {
-		_msg = msg;
-		if (len > 0) {
-			_data = new byte[len];
-			memcpy(_data, data, len);
-		}
-		_len = len;
-		_next = NULL;
-	}
-
-	MidiEvent_MT32() {
-		if (_len > 0)
-			delete _data;
-	}
-};
-
-class MidiDriver_ThreadedMT32 : public MidiDriver_MT32 {
-private:
-	OSystem::Mutex _eventMutex;
-	MidiEvent_MT32 *_events;
-	TimerManager::TimerProc _timer_proc;
-
-	void pushMidiEvent(MidiEvent_MT32 *event);
-	MidiEvent_MT32 *popMidiEvent();
-
-protected:
-	void send(uint32 b);
-	void sysEx(const byte *msg, uint16 length);
-
-public:
-	MidiDriver_ThreadedMT32(Audio::Mixer *mixer);
-
-	void onTimer();
-	void close();
-	void setTimerCallback(void *timer_param, TimerManager::TimerProc timer_proc);
-};
-
-
-MidiDriver_ThreadedMT32::MidiDriver_ThreadedMT32(Audio::Mixer *mixer) : MidiDriver_MT32(mixer) {
-	_events = NULL;
-	_timer_proc = NULL;
-}
-
-void MidiDriver_ThreadedMT32::close() {
-	MidiDriver_MT32::close();
-	while ((popMidiEvent() != NULL)) {
-		// Just eat any leftover events
-	}
-}
-
-void MidiDriver_ThreadedMT32::setTimerCallback(void *timer_param, TimerManager::TimerProc timer_proc) {
-	if (!_timer_proc || !timer_proc) {
-		if (_timer_proc)
-			_vm->_timer->removeTimerProc(_timer_proc);
-		_timer_proc = timer_proc;
-		if (timer_proc)
-			_vm->_timer->installTimerProc(timer_proc, getBaseTempo(), timer_param);
-	}
-}
-
-void MidiDriver_ThreadedMT32::pushMidiEvent(MidiEvent_MT32 *event) {
-	Common::StackLock lock(_eventMutex);
-	if (_events == NULL) {
-		_events = event;
-	} else {
-		MidiEvent_MT32 *last = _events;
-		while (last->_next != NULL)
-			last = last->_next;
-		last->_next = event;
-	}
-}
-
-MidiEvent_MT32 *MidiDriver_ThreadedMT32::popMidiEvent() {
-	Common::StackLock lock(_eventMutex);
-	MidiEvent_MT32 *event;
-	event = _events;
-	if (event != NULL)
-		_events = event->_next;
-	return event;
-}
-
-void MidiDriver_ThreadedMT32::send(uint32 b) {
-	MidiEvent_MT32 *event = new MidiEvent_MT32(b, NULL, 0);
-	pushMidiEvent(event);
-}
-
-void MidiDriver_ThreadedMT32::sysEx(const byte *msg, uint16 length) {
-	MidiEvent_MT32 *event = new MidiEvent_MT32(0xFFFFFFFF, msg, length);
-	pushMidiEvent(event);
-}
-
-void MidiDriver_ThreadedMT32::onTimer() {
-	MidiEvent_MT32 *event;
-	while ((event = popMidiEvent()) != NULL) {
-		if (event->_msg == 0xFFFFFFFF) {
-			MidiDriver_MT32::sysEx(event->_data, event->_len);
-		} else {
-			MidiDriver_MT32::send(event->_msg);
-		}
-		delete event;
-	}
-}
-#endif
-
-
-// Plugin interface
-
-class MT32EmuMusicPlugin : public MusicPluginObject {
-public:
-	const char *getName() const {
-		return _s("MT-32 Emulator");
-	}
-
-	const char *getId() const {
-		return "mt32";
-	}
-
-	MusicDevices getDevices() const;
-	Common::Error createInstance(MidiDriver **mididriver, MidiDriver::DeviceHandle = 0) const;
-};
-
-MusicDevices MT32EmuMusicPlugin::getDevices() const {
-	MusicDevices devices;
-	devices.push_back(MusicDevice(this, "", MT_MT32));
-	return devices;
-}
-
-Common::Error MT32EmuMusicPlugin::createInstance(MidiDriver **mididriver, MidiDriver::DeviceHandle) const {
-	if (ConfMan.hasKey("extrapath"))
-		SearchMan.addDirectory("extrapath", ConfMan.get("extrapath"));
-
-	*mididriver = new MidiDriver_MT32(g_system->getMixer());
-
-	return Common::kNoError;
-}
-
-//#if PLUGIN_ENABLED_DYNAMIC(MT32)
-	//REGISTER_PLUGIN_DYNAMIC(MT32, PLUGIN_TYPE_MUSIC, MT32EmuMusicPlugin);
-//#else
-	REGISTER_PLUGIN_STATIC(MT32, PLUGIN_TYPE_MUSIC, MT32EmuMusicPlugin);
-//#endif
-
-#endif
diff --git a/sound/softsynth/mt32/freeverb.cpp b/sound/softsynth/mt32/freeverb.cpp
deleted file mode 100644
index 4ee6e20117..0000000000
--- a/sound/softsynth/mt32/freeverb.cpp
+++ /dev/null
@@ -1,310 +0,0 @@
-/* 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.
- *
- * $URL$
- * $Id$
- *
- */
-
-// Comb filter implementation
-//
-// Written by
-// http://www.dreampoint.co.uk
-// This code is public domain
-
-#include "sound/softsynth/mt32/freeverb.h"
-
-comb::comb() {
-	filterstore = 0;
-	bufidx = 0;
-}
-
-void comb::setbuffer(float *buf, int size) {
-	buffer = buf;
-	bufsize = size;
-}
-
-void comb::mute() {
-	for (int i = 0; i < bufsize; i++)
-		buffer[i] = 0;
-}
-
-void comb::setdamp(float val) {
-	damp1 = val;
-	damp2 = 1 - val;
-}
-
-float comb::getdamp() {
-	return damp1;
-}
-
-void comb::setfeedback(float val) {
-	feedback = val;
-}
-
-float comb::getfeedback() {
-	return feedback;
-}
-
-// Allpass filter implementation
-
-allpass::allpass() {
-	bufidx = 0;
-}
-
-void allpass::setbuffer(float *buf, int size) {
-	buffer = buf;
-	bufsize = size;
-}
-
-void allpass::mute() {
-	for (int i = 0; i < bufsize; i++)
-		buffer[i] = 0;
-}
-
-void allpass::setfeedback(float val) {
-	feedback = val;
-}
-
-float allpass::getfeedback() {
-	return feedback;
-}
-
-// Reverb model implementation
-
-revmodel::revmodel() {
-	// Tie the components to their buffers
-	combL[0].setbuffer(bufcombL1,combtuningL1);
-	combR[0].setbuffer(bufcombR1,combtuningR1);
-	combL[1].setbuffer(bufcombL2,combtuningL2);
-	combR[1].setbuffer(bufcombR2,combtuningR2);
-	combL[2].setbuffer(bufcombL3,combtuningL3);
-	combR[2].setbuffer(bufcombR3,combtuningR3);
-	combL[3].setbuffer(bufcombL4,combtuningL4);
-	combR[3].setbuffer(bufcombR4,combtuningR4);
-	combL[4].setbuffer(bufcombL5,combtuningL5);
-	combR[4].setbuffer(bufcombR5,combtuningR5);
-	combL[5].setbuffer(bufcombL6,combtuningL6);
-	combR[5].setbuffer(bufcombR6,combtuningR6);
-	combL[6].setbuffer(bufcombL7,combtuningL7);
-	combR[6].setbuffer(bufcombR7,combtuningR7);
-	combL[7].setbuffer(bufcombL8,combtuningL8);
-	combR[7].setbuffer(bufcombR8,combtuningR8);
-	allpassL[0].setbuffer(bufallpassL1,allpasstuningL1);
-	allpassR[0].setbuffer(bufallpassR1,allpasstuningR1);
-	allpassL[1].setbuffer(bufallpassL2,allpasstuningL2);
-	allpassR[1].setbuffer(bufallpassR2,allpasstuningR2);
-	allpassL[2].setbuffer(bufallpassL3,allpasstuningL3);
-	allpassR[2].setbuffer(bufallpassR3,allpasstuningR3);
-	allpassL[3].setbuffer(bufallpassL4,allpasstuningL4);
-	allpassR[3].setbuffer(bufallpassR4,allpasstuningR4);
-
-	// Set default values
-	allpassL[0].setfeedback(0.5f);
-	allpassR[0].setfeedback(0.5f);
-	allpassL[1].setfeedback(0.5f);
-	allpassR[1].setfeedback(0.5f);
-	allpassL[2].setfeedback(0.5f);
-	allpassR[2].setfeedback(0.5f);
-	allpassL[3].setfeedback(0.5f);
-	allpassR[3].setfeedback(0.5f);
-	setmode(initialmode);
-	setwet(initialwet);
-	setroomsize(initialroom);
-	setdry(initialdry);
-	setdamp(initialdamp);
-	setwidth(initialwidth);
-
-	// Buffer will be full of rubbish - so we MUST mute them
-	mute();
-}
-
-void revmodel::mute() {
-	int i;
-
-	if (getmode() >= freezemode)
-		return;
-
-	for (i = 0; i < numcombs; i++) {
-		combL[i].mute();
-		combR[i].mute();
-	}
-
-	for (i = 0; i < numallpasses; i++) {
-		allpassL[i].mute();
-		allpassR[i].mute();
-	}
-}
-
-void revmodel::processreplace(float *inputL, float *inputR, float *outputL, float *outputR, long numsamples, int skip) {
-	float outL, outR, input;
-
-	while (numsamples-- > 0) {
-		int i;
-
-		outL = outR = 0;
-		input = (*inputL + *inputR) * gain;
-
-		// Accumulate comb filters in parallel
-		for (i = 0; i < numcombs; i++) {
-			outL += combL[i].process(input);
-			outR += combR[i].process(input);
-		}
-
-		// Feed through allpasses in series
-		for (i = 0; i < numallpasses; i++) {
-			outL = allpassL[i].process(outL);
-			outR = allpassR[i].process(outR);
-		}
-
-		// Calculate output REPLACING anything already there
-		*outputL = outL * wet1 + outR * wet2 + *inputL * dry;
-		*outputR = outR * wet1 + outL * wet2 + *inputR * dry;
-
-		// Increment sample pointers, allowing for interleave (if any)
-		inputL += skip;
-		inputR += skip;
-		outputL += skip;
-		outputR += skip;
-	}
-}
-
-void revmodel::processmix(float *inputL, float *inputR, float *outputL, float *outputR, long numsamples, int skip) {
-	float outL, outR, input;
-
-	while (numsamples-- > 0) {
-		int i;
-
-		outL = outR = 0;
-		input = (*inputL + *inputR) * gain;
-
-		// Accumulate comb filters in parallel
-		for (i = 0; i < numcombs; i++) {
-			outL += combL[i].process(input);
-			outR += combR[i].process(input);
-		}
-
-		// Feed through allpasses in series
-		for (i = 0; i < numallpasses; i++) {
-			outL = allpassL[i].process(outL);
-			outR = allpassR[i].process(outR);
-		}
-
-		// Calculate output MIXING with anything already there
-		*outputL += outL * wet1 + outR * wet2 + *inputL * dry;
-		*outputR += outR * wet1 + outL * wet2 + *inputR * dry;
-
-		// Increment sample pointers, allowing for interleave (if any)
-		inputL += skip;
-		inputR += skip;
-		outputL += skip;
-		outputR += skip;
-	}
-}
-
-void revmodel::update() {
-	// Recalculate internal values after parameter change
-
-	int i;
-
-	wet1 = wet * (width / 2 + 0.5f);
-	wet2 = wet * ((1 - width) / 2);
-
-	if (mode >= freezemode) {
-		roomsize1 = 1;
-		damp1 = 0;
-		gain = muted;
-	} else {
-		roomsize1 = roomsize;
-		damp1 = damp;
-		gain = fixedgain;
-	}
-
-	for (i = 0; i < numcombs; i++) {
-		combL[i].setfeedback(roomsize1);
-		combR[i].setfeedback(roomsize1);
-	}
-
-	for (i = 0; i < numcombs; i++) {
-		combL[i].setdamp(damp1);
-		combR[i].setdamp(damp1);
-	}
-}
-
-// The following get/set functions are not inlined, because
-// speed is never an issue when calling them, and also
-// because as you develop the reverb model, you may
-// wish to take dynamic action when they are called.
-
-void revmodel::setroomsize(float value) {
-	roomsize = (value * scaleroom) + offsetroom;
-	update();
-}
-
-float revmodel::getroomsize() {
-	return (roomsize - offsetroom) / scaleroom;
-}
-
-void revmodel::setdamp(float value) {
-	damp = value * scaledamp;
-	update();
-}
-
-float revmodel::getdamp() {
-	return damp / scaledamp;
-}
-
-void revmodel::setwet(float value) {
-	wet = value * scalewet;
-	update();
-}
-
-float revmodel::getwet() {
-	return wet / scalewet;
-}
-
-void revmodel::setdry(float value) {
-	dry = value * scaledry;
-}
-
-float revmodel::getdry() {
-	return dry / scaledry;
-}
-
-void revmodel::setwidth(float value) {
-	width = value;
-	update();
-}
-
-float revmodel::getwidth() {
-	return width;
-}
-
-void revmodel::setmode(float value) {
-	mode = value;
-	update();
-}
-
-float revmodel::getmode() {
-	if (mode >= freezemode)
-		return 1;
-	else
-		return 0;
-}
diff --git a/sound/softsynth/mt32/freeverb.h b/sound/softsynth/mt32/freeverb.h
deleted file mode 100644
index 8310aca3e3..0000000000
--- a/sound/softsynth/mt32/freeverb.h
+++ /dev/null
@@ -1,244 +0,0 @@
-/* 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.
- *
- * $URL$
- * $Id$
- *
- */
-
-// Macro for killing denormalled numbers
-//
-// Written by Jezar at Dreampoint, June 2000
-// http://www.dreampoint.co.uk
-// Based on IS_DENORMAL macro by Jon Watte
-// This code is public domain
-
-#ifndef FREEVERB_H
-#define FREEVERB_H
-
-// FIXME: Fix this really ugly hack
-inline float undenormalise(void *sample) {
-	if (((*(unsigned int*)sample) &  0x7f800000) == 0)
-		return 0.0f;
-	return *(float*)sample;
-}
-
-// Comb filter class declaration
-
-class comb {
-public:
-	comb();
-	void setbuffer(float *buf, int size);
-	inline float process(float inp);
-	void mute();
-	void setdamp(float val);
-	float getdamp();
-	void setfeedback(float val);
-	float getfeedback();
-private:
-	float feedback;
-	float filterstore;
-	float damp1;
-	float damp2;
-	float *buffer;
-	int bufsize;
-	int bufidx;
-};
-
-
-// Big to inline - but crucial for speed
-
-inline float comb::process(float input) {
-	float output;
-
-	output = buffer[bufidx];
-	undenormalise(&output);
-
-	filterstore = (output * damp2) + (filterstore * damp1);
-	undenormalise(&filterstore);
-
-	buffer[bufidx] = input + (filterstore * feedback);
-
-	if (++bufidx >= bufsize)
-		bufidx = 0;
-
-	return output;
-}
-
-// Allpass filter declaration
-
-class allpass {
-public:
-	allpass();
-	void setbuffer(float *buf, int size);
-	inline float process(float inp);
-	void mute();
-	void setfeedback(float val);
-	float getfeedback();
-private:
-	float feedback;
-	float *buffer;
-	int bufsize;
-	int bufidx;
-};
-
-
-// Big to inline - but crucial for speed
-
-inline float allpass::process(float input) {
-	float output;
-	float bufout;
-
-	bufout = buffer[bufidx];
-	undenormalise(&bufout);
-
-	output = -input + bufout;
-	buffer[bufidx] = input + (bufout * feedback);
-
-	if (++bufidx >= bufsize)
-		bufidx = 0;
-
-	return output;
-}
-
-
-// Reverb model tuning values
-
-const int	numcombs	= 8;
-const int	numallpasses	= 4;
-const float	muted		= 0;
-const float	fixedgain	= 0.015f;
-const float	scalewet	= 3;
-const float	scaledry	= 2;
-const float	scaledamp	= 0.4f;
-const float	scaleroom	= 0.28f;
-const float	offsetroom	= 0.7f;
-const float	initialroom	= 0.5f;
-const float	initialdamp	= 0.5f;
-const float	initialwet	= 1 / scalewet;
-const float	initialdry	= 0;
-const float	initialwidth	= 1;
-const float	initialmode	= 0;
-const float	freezemode	= 0.5f;
-const int	stereospread	= 23;
-
-// These values assume 44.1KHz sample rate
-// they will probably be OK for 48KHz sample rate
-// but would need scaling for 96KHz (or other) sample rates.
-// The values were obtained by listening tests.
-const int combtuningL1		= 1116;
-const int combtuningR1		= 1116 + stereospread;
-const int combtuningL2		= 1188;
-const int combtuningR2		= 1188 + stereospread;
-const int combtuningL3		= 1277;
-const int combtuningR3		= 1277 + stereospread;
-const int combtuningL4		= 1356;
-const int combtuningR4		= 1356 + stereospread;
-const int combtuningL5		= 1422;
-const int combtuningR5		= 1422 + stereospread;
-const int combtuningL6		= 1491;
-const int combtuningR6		= 1491 + stereospread;
-const int combtuningL7		= 1557;
-const int combtuningR7		= 1557 + stereospread;
-const int combtuningL8		= 1617;
-const int combtuningR8		= 1617 + stereospread;
-const int allpasstuningL1	= 556;
-const int allpasstuningR1	= 556 + stereospread;
-const int allpasstuningL2	= 441;
-const int allpasstuningR2	= 441 + stereospread;
-const int allpasstuningL3	= 341;
-const int allpasstuningR3	= 341 + stereospread;
-const int allpasstuningL4	= 225;
-const int allpasstuningR4	= 225 + stereospread;
-
-
-// Reverb model declaration
-
-class revmodel {
-public:
-	revmodel();
-	void mute();
-	void processmix(float *inputL, float *inputR, float *outputL, float *outputR, long numsamples, int skip);
-	void processreplace(float *inputL, float *inputR, float *outputL, float *outputR, long numsamples, int skip);
-	void setroomsize(float value);
-	float getroomsize();
-	void setdamp(float value);
-	float getdamp();
-	void setwet(float value);
-	float getwet();
-	void setdry(float value);
-	float getdry();
-	void setwidth(float value);
-	float getwidth();
-	void setmode(float value);
-	float getmode();
-private:
-	void update();
-
-	float gain;
-	float roomsize, roomsize1;
-	float damp, damp1;
-	float wet, wet1, wet2;
-	float dry;
-	float width;
-	float mode;
-
-	// The following are all declared inline
-	// to remove the need for dynamic allocation
-	// with its subsequent error-checking messiness
-
-	// Comb filters
-	comb combL[numcombs];
-	comb combR[numcombs];
-
-	// Allpass filters
-	allpass	allpassL[numallpasses];
-	allpass	allpassR[numallpasses];
-
-	// Buffers for the combs
-	float bufcombL1[combtuningL1];
-	float bufcombR1[combtuningR1];
-	float bufcombL2[combtuningL2];
-	float bufcombR2[combtuningR2];
-	float bufcombL3[combtuningL3];
-	float bufcombR3[combtuningR3];
-	float bufcombL4[combtuningL4];
-	float bufcombR4[combtuningR4];
-	float bufcombL5[combtuningL5];
-	float bufcombR5[combtuningR5];
-	float bufcombL6[combtuningL6];
-	float bufcombR6[combtuningR6];
-	float bufcombL7[combtuningL7];
-	float bufcombR7[combtuningR7];
-	float bufcombL8[combtuningL8];
-	float bufcombR8[combtuningR8];
-
-	// Buffers for the allpasses
-	float bufallpassL1[allpasstuningL1];
-	float bufallpassR1[allpasstuningR1];
-	float bufallpassL2[allpasstuningL2];
-	float bufallpassR2[allpasstuningR2];
-	float bufallpassL3[allpasstuningL3];
-	float bufallpassR3[allpasstuningR3];
-	float bufallpassL4[allpasstuningL4];
-	float bufallpassR4[allpasstuningR4];
-};
-
-#endif
diff --git a/sound/softsynth/mt32/i386.cpp b/sound/softsynth/mt32/i386.cpp
deleted file mode 100644
index f092189d76..0000000000
--- a/sound/softsynth/mt32/i386.cpp
+++ /dev/null
@@ -1,849 +0,0 @@
-/* Copyright (c) 2003-2005 Various contributors
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to
- * deal in the Software without restriction, including without limitation the
- * rights to use, copy, modify, merge, publish, distribute, sublicense, and/or
- * sell copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
- * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
- * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS
- * IN THE SOFTWARE.
- */
-
-#include "mt32emu.h"
-
-#ifdef MT32EMU_HAVE_X86
-
-namespace MT32Emu {
-
-#ifndef _MSC_VER
-
-#define eflag(value) __asm__ __volatile__("pushfl \n popfl \n" : : "a"(value))
-#define cpuid_flag  (1 << 21)
-
-static inline bool atti386_DetectCPUID() {
-	unsigned int result;
-
-	// Is there a cpuid?
-	result = cpuid_flag; // set test
-	eflag(result);
-	if (!(result & cpuid_flag))
-		return false;
-
-	result = 0; // clear test
-	eflag(result);
-	if (result & cpuid_flag)
-		return false;
-
-	return true;
-}
-
-static inline bool atti386_DetectSIMD() {
-	unsigned int result;
-
-	if (atti386_DetectCPUID() == false)
-		return false;
-
-	/* check cpuid */
-	__asm__ __volatile__(
-		"pushl  %%ebx            \n" \
-		"movl   $1, %%eax        \n" \
-		"cpuid                   \n" \
-		"movl   %%edx, %0        \n" \
-		"popl   %%ebx            \n" \
-		: "=r"(result) : : "eax", "ecx", "edx");
-
-	if (result & (1 << 25))
-		return true;
-
-	return false;
-}
-
-static inline bool atti386_Detect3DNow() {
-	unsigned int result;
-
-	if (atti386_DetectCPUID() == false)
-		return false;
-
-	// get cpuid
-	__asm__ __volatile__(
-		"pushl  %%ebx            \n" \
-		"movl   $0x80000001, %%eax \n" \
-		"cpuid                     \n" \
-		"movl   %%edx, %0          \n" \
-		"popl   %%ebx            \n" \
-		: "=r"(result) : : "eax", "ecx", "edx");
-
-	if (result & 0x80000000)
-		return true;
-
-	return false;
-}
-
-
-static inline float atti386_iir_filter_sse(float *output, float *hist1_ptr, float *coef_ptr) {
-	__asm__  __volatile__ (
-		"pushl  %1                      \n" \
-		"pushl  %2                      \n" \
-		"movss  0(%0), %%xmm1           \n" \
-		"movups 0(%1), %%xmm2           \n" \
-		"movlps 0(%2), %%xmm3           \n" \
-		"                               \n" \
-		"shufps $0x44, %%xmm3, %%xmm3   \n" \
-		"                               \n" \
-		"mulps  %%xmm3, %%xmm2          \n" \
-		"                               \n" \
-		"subss  %%xmm2, %%xmm1          \n" \
-		"shufps $0x39,  %%xmm2, %%xmm2  \n" \
-		"subss  %%xmm2, %%xmm1          \n" \
-		"                               \n" \
-		"movss  %%xmm1, 0(%2)           \n" \
-		"                               \n" \
-		"shufps $0x39,  %%xmm2, %%xmm2  \n" \
-		"addss  %%xmm2, %%xmm1          \n" \
-		"                               \n" \
-		"shufps $0x39,  %%xmm2, %%xmm2  \n" \
-		"addss  %%xmm2, %%xmm1          \n" \
-		"                               \n" \
-		"movss  %%xmm3, 4(%2)           \n" \
-		"                               \n" \
-		"addl   $16, %1                 \n" \
-		"addl   $8, %2                  \n" \
-		"                               \n" \
-		"movups 0(%1), %%xmm2           \n" \
-		"                               \n" \
-		"movlps 0(%2), %%xmm3           \n" \
-		"shufps $0x44, %%xmm3, %%xmm3   \n" \
-		"                               \n" \
-		"mulps %%xmm3, %%xmm2           \n" \
-		"                               \n" \
-		"subss  %%xmm2, %%xmm1          \n" \
-		"shufps $0x39,  %%xmm2, %%xmm2  \n" \
-		"subss  %%xmm2, %%xmm1          \n" \
-		"                               \n" \
-		"movss %%xmm1, 0(%2)            \n" \
-		"                               \n" \
-		"shufps $0x39, %%xmm2, %%xmm2   \n" \
-		"addss %%xmm2, %%xmm1           \n" \
-		"                               \n" \
-		"shufps $0x39, %%xmm2, %%xmm2   \n" \
-		"addss %%xmm2, %%xmm1           \n" \
-		"                               \n" \
-		"movss %%xmm3, 4(%2)            \n" \
-		"movss %%xmm1, 0(%0)            \n" \
-		"popl %2                        \n" \
-		"popl %1                        \n" \
-		: : "r"(output), "r"(coef_ptr), "r"(hist1_ptr)
-		: "memory"
-#ifdef __SSE__
-		, "xmm1", "xmm2", "xmm3"
-#endif
-		);
-
-	return *output;
-}
-
-static inline float atti386_iir_filter_3DNow(float output, float *hist1_ptr, float *coef_ptr) {
-	float tmp;
-
-	__asm__ __volatile__ (
-		"movq %0, %%mm1       \n" \
-		"                     \n" \
-		"movl  %1, %%edi      \n" \
-		"movq 0(%%edi), %%mm2 \n" \
-		"                     \n" \
-		"movl %2, %%eax;      \n" \
-		"movq 0(%%eax), %%mm3 \n" \
-		"                     \n" \
-		"pfmul %%mm3, %%mm2   \n" \
-		"pfsub %%mm2, %%mm1   \n" \
-		"                     \n" \
-		"psrlq $32, %%mm2     \n" \
-		"pfsub %%mm2, %%mm1   \n" \
-		"                     \n" \
-		"movd %%mm1, %3       \n" \
-		"                     \n" \
-		"addl  $8, %%edi      \n" \
-		"movq 0(%%edi), %%mm2 \n" \
-		"movq 0(%%eax), %%mm3 \n" \
-		"                     \n" \
-		"pfmul %%mm3, %%mm2   \n" \
-		"pfadd %%mm2, %%mm1   \n" \
-		"                     \n" \
-		"psrlq $32, %%mm2     \n" \
-		"pfadd %%mm2, %%mm1   \n" \
-		"                     \n" \
-		"pushl %3             \n" \
-		"popl 0(%%eax)        \n" \
-		"                     \n" \
-		"movd %%mm3, 4(%%eax) \n" \
-		"                     \n" \
-		"addl $8, %%edi       \n" \
-		"addl $8, %%eax       \n" \
-		"                     \n" \
-		"movq 0(%%edi), %%mm2 \n" \
-		"movq 0(%%eax), %%mm3 \n" \
-		"                     \n" \
-		"pfmul %%mm3, %%mm2   \n" \
-		"pfsub %%mm2, %%mm1   \n" \
-		"                     \n" \
-		"psrlq $32, %%mm2     \n" \
-		"pfsub %%mm2, %%mm1   \n" \
-		"                     \n" \
-		"movd %%mm1, %3       \n" \
-		"                     \n" \
-		"addl $8, %%edi       \n" \
-		"movq 0(%%edi), %%mm2 \n" \
-		"movq 0(%%eax), %%mm3 \n" \
-		"                     \n" \
-		"pfmul %%mm3, %%mm2   \n" \
-		"pfadd %%mm2, %%mm1   \n" \
-		"                     \n" \
-		"psrlq $32, %%mm2     \n" \
-		"pfadd %%mm2, %%mm1   \n" \
-		"                     \n" \
-		"pushl %3             \n" \
-		"popl 0(%%eax)        \n" \
-		"movd %%mm3, 4(%%eax) \n" \
-		"                     \n" \
-		"movd %%mm1, %0       \n" \
-		"femms                \n" \
-		: "=m"(output) : "g"(coef_ptr), "g"(hist1_ptr), "m"(tmp)
-		: "eax", "edi", "memory"
-#ifdef __MMX__
-		, "mm1", "mm2", "mm3"
-#endif
-		);
-
-	return output;
-}
-
-static inline void atti386_produceOutput1(int tmplen, Bit16s myvolume, Bit16s *useBuf, Bit16s *snd) {
-	__asm__ __volatile__(
-		"movl %0,  %%ecx      \n" \
-		"movw %1,  %%ax       \n" \
-		"shll $16, %%eax      \n" \
-		"movw %1,  %%ax       \n" \
-		"movd %%eax, %%mm3    \n" \
-		"movd %%eax, %%mm2    \n" \
-		"psllq $32, %%mm3     \n" \
-		"por %%mm2, %%mm3     \n" \
-		"movl %2, %%esi       \n" \
-		"movl %3, %%edi       \n" \
-		"1:                   \n" \
-		"movq 0(%%esi), %%mm1 \n" \
-		"movq 0(%%edi), %%mm2 \n" \
-		"pmulhw %%mm3, %%mm1  \n" \
-		"paddw %%mm2, %%mm1   \n" \
-		"movq %%mm1, 0(%%edi) \n" \
-		"                     \n" \
-		"addl $8, %%esi       \n" \
-		"addl $8, %%edi       \n" \
-		"                     \n" \
-		"decl %%ecx           \n" \
-		"cmpl $0, %%ecx       \n" \
-		"jg   1b              \n" \
-		"emms                 \n" \
-		: : "g"(tmplen), "g"(myvolume), "g"(useBuf), "g"(snd)
-		: "eax", "ecx", "edi", "esi", "memory"
-#ifdef __MMX__
-		, "mm1", "mm2", "mm3"
-#endif
-		);
-}
-
-static inline void atti386_produceOutput2(Bit32u len, Bit16s *snd, float *sndbufl, float *sndbufr, float *multFactor) {
-	__asm__ __volatile__(
-		"movl  %4, %%ecx         \n" \
-		"shrl  $1, %%ecx         \n" \
-		"addl  $4, %%ecx         \n" \
-		"pushl %%ecx             \n" \
-		"                        \n" \
-		"movl %0, %%esi          \n" \
-		"movups 0(%%esi), %%xmm1 \n" \
-		"                        \n" \
-		"movl %1, %%esi          \n" \
-		"movl %2, %%edi          \n" \
-		"1:                      \n" \
-		"xorl %%eax, %%eax       \n" \
-		"movw 0(%1), %%ax        \n" \
-		"cwde                    \n" \
-		"incl %1                 \n" \
-		"incl %1                 \n" \
-		"movd  %%eax, %%mm1      \n" \
-		"psrlq $32, %%mm1        \n" \
-		"movw 0(%1), %%ax        \n" \
-		"incl %1                 \n" \
-		"incl %1                 \n" \
-		"movd %%eax, %%mm2       \n" \
-		"por %%mm2, %%mm1        \n" \
-		"                        \n" \
-		"decl %%ecx              \n" \
-		"jnz 1b                  \n" \
-		"                        \n" \
-		"popl %%ecx              \n" \
-		"movl %1, %%esi          \n" \
-		"movl %3, %%edi          \n" \
-		"incl %%esi              \n" \
-		"2:                      \n" \
-		"decl %%ecx              \n" \
-		"jnz 2b                  \n" \
-		: : "g"(multFactor), "r"(snd), "g"(sndbufl), "g"(sndbufr), "g"(len)
-		: "eax", "ecx", "edi", "esi", "mm1", "mm2", "xmm1", "memory");
-}
-
-static inline void atti386_mixBuffers(Bit16s * buf1, Bit16s *buf2, int len) {
-	__asm__ __volatile__(
-		"movl %0, %%ecx       \n" \
-		"movl %1, %%esi       \n" \
-		"movl %2, %%edi       \n" \
-		"1:                   \n" \
-		"movq 0(%%edi), %%mm1 \n" \
-		"movq 0(%%esi), %%mm2 \n" \
-		"paddw %%mm2, %%mm1   \n" \
-		"movq %%mm1, 0(%%esi) \n" \
-		"addl $8, %%edi       \n" \
-		"addl $8, %%esi       \n" \
-		"decl %%ecx           \n" \
-		"cmpl $0, %%ecx       \n" \
-		"jg   1b              \n" \
-		"emms                 \n" \
-		: : "g"(len), "g"(buf1), "g"(buf2)
-		: "ecx", "edi", "esi", "memory"
-#ifdef __MMX__
-		, "mm1", "mm2"
-#endif
-		);
-}
-
-static inline void atti386_mixBuffersRingMix(Bit16s * buf1, Bit16s *buf2, int len) {
-	__asm__ __volatile__(
-		"movl %0, %%ecx       \n" \
-		"movl %1, %%esi       \n" \
-		"movl %2, %%edi       \n" \
-		"1:                   \n" \
-		"movq 0(%%esi), %%mm1 \n" \
-		"movq 0(%%edi), %%mm2 \n" \
-		"movq %%mm1, %%mm3    \n" \
-		"pmulhw %%mm2, %%mm1  \n" \
-		"paddw %%mm3, %%mm1   \n" \
-		"movq %%mm1, 0(%%esi) \n" \
-		"addl $8, %%edi       \n" \
-		"addl $8, %%esi       \n" \
-		"decl %%ecx           \n" \
-		"cmpl $0, %%ecx       \n" \
-		"jg   1b              \n" \
-		"emms                 \n" \
-		: : "g"(len), "g"(buf1), "g"(buf2)
-		: "ecx", "edi", "esi", "memory"
-#ifdef __MMX__
-		, "mm1", "mm2", "mm3"
-#endif
-		);
-}
-
-static inline void atti386_mixBuffersRing(Bit16s * buf1, Bit16s *buf2, int len) {
-	__asm__ __volatile__(
-		"movl %0, %%ecx       \n" \
-		"movl %1, %%esi       \n" \
-		"movl %2, %%edi       \n" \
-		"1:                   \n" \
-		"movq 0(%%esi), %%mm1 \n" \
-		"movq 0(%%edi), %%mm2 \n" \
-		"pmulhw %%mm2, %%mm1  \n" \
-		"movq %%mm1, 0(%%esi) \n" \
-		"addl $8, %%edi       \n" \
-		"addl $8, %%esi       \n" \
-		"decl %%ecx           \n" \
-		"cmpl $0, %%ecx       \n" \
-		"jg   1b              \n" \
-		"emms                 \n" \
-		: : "g"(len), "g"(buf1), "g"(buf2)
-		: "ecx", "edi", "esi", "memory"
-#ifdef __MMX__
-		, "mm1", "mm2"
-#endif
-		);
-}
-
-static inline void atti386_partialProductOutput(int quadlen, Bit16s leftvol, Bit16s rightvol, Bit16s *partialBuf, Bit16s *p1buf) {
-	__asm__ __volatile__(
-		"movl %0, %%ecx       \n" \
-		"movw %1, %%ax        \n" \
-		"shll $16, %%eax      \n" \
-		"movw %2, %%ax        \n" \
-		"movd %%eax, %%mm1    \n" \
-		"movd %%eax, %%mm2    \n" \
-		"psllq $32, %%mm1     \n" \
-		"por  %%mm2, %%mm1    \n" \
-		"movl %3, %%edi       \n" \
-		"movl %4, %%esi       \n" \
-	    "pushl %%ebx          \n" \
-		"1:                   \n" \
-		"movw 0(%%esi), %%bx  \n" \
-		"addl $2, %%esi       \n" \
-		"movw 0(%%esi), %%dx  \n" \
-		"addl $2, %%esi       \n" \
-		""                        \
-		"movw %%dx, %%ax      \n" \
-		"shll $16, %%eax      \n" \
-		"movw %%dx, %%ax      \n" \
-		"movd %%eax, %%mm2    \n" \
-		"psllq $32, %%mm2     \n" \
-		"movw %%bx, %%ax      \n" \
-		"shll $16, %%eax      \n" \
-		"movw %%bx, %%ax      \n" \
-		"movd %%eax, %%mm3    \n" \
-		"por  %%mm3, %%mm2    \n" \
-		""                        \
-		"pmulhw %%mm1, %%mm2  \n" \
-		"movq %%mm2, 0(%%edi) \n" \
-		"addl $8, %%edi       \n" \
-		""                        \
-		"decl %%ecx           \n" \
-		"cmpl $0, %%ecx       \n" \
-		"jg 1b                \n" \
-		"emms                 \n"  \
-	    "popl %%ebx           \n" \
-		: : "g"(quadlen), "g"(leftvol), "g"(rightvol), "g"(partialBuf), "g"(p1buf)
-		: "eax", "ecx", "edx", "edi", "esi", "memory"
-#ifdef __MMX__
-		, "mm1", "mm2", "mm3"
-#endif
-		);
-}
-
-#endif
-
-bool DetectSIMD() {
-#ifdef _MSC_VER
-	bool found_simd;
-	__asm {
-		pushfd
-		pop eax // get EFLAGS into eax
-		mov ebx,eax // keep a copy
-		xor eax,0x200000
-		// toggle CPUID bit
-
-		push eax
-		popfd // set new EFLAGS
-		pushfd
-		pop eax // EFLAGS back into eax
-
-		xor eax,ebx
-		// have we changed the ID bit?
-
-		je NO_SIMD
-		// No, no CPUID instruction
-
-		// we could toggle the
-		// ID bit so CPUID is present
-		mov eax,1
-
-		cpuid // get processor features
-		test edx,1<<25 // check the SIMD bit
-		jz NO_SIMD
-		mov found_simd,1
-		jmp DONE
-		NO_SIMD:
-		mov found_simd,0
-		DONE:
-	}
-	return found_simd;
-#else
-	return atti386_DetectSIMD();
-#endif
-}
-
-bool Detect3DNow() {
-#ifdef _MSC_VER
-	bool found3D = false;
-	__asm {
-		pushfd
-		pop eax
-		mov edx, eax
-		xor eax, 00200000h
-		push eax
-		popfd
-		pushfd
-		pop eax
-		xor eax, edx
-		jz NO_3DNOW
-
-		mov eax, 80000000h
-		cpuid
-
-		cmp eax, 80000000h
-		jbe NO_3DNOW
-
-		mov eax, 80000001h
-		cpuid
-		test edx, 80000000h
-		jz NO_3DNOW
-		mov found3D, 1
-NO_3DNOW:
-
-	}
-	return found3D;
-#else
-	return atti386_Detect3DNow();
-#endif
-}
-
-float iir_filter_sse(float input,float *hist1_ptr, float *coef_ptr) {
-	float output;
-
-	// 1st number of coefficients array is overall input scale factor, or filter gain
-	output = input * (*coef_ptr++);
-
-#ifdef _MSC_VER
-	__asm {
-
-		movss xmm1,	output
-
-		mov eax, coef_ptr
-		movups xmm2, [eax]
-
-		mov eax, hist1_ptr
-		movlps xmm3, [eax]
-		shufps xmm3, xmm3, 44h
-		// hist1_ptr+1, hist1_ptr, hist1_ptr+1, hist1_ptr
-
-		mulps xmm2, xmm3
-
-		subss xmm1, xmm2
-		// Rotate elements right
-		shufps xmm2, xmm2, 39h
-		subss xmm1, xmm2
-
-		// Store new_hist
-		movss DWORD PTR [eax], xmm1
-
-		// Rotate elements right
-		shufps xmm2, xmm2, 39h
-		addss xmm1, xmm2
-
-		// Rotate elements right
-		shufps xmm2, xmm2, 39h
-		addss xmm1, xmm2
-
-		// Store previous hist
-		movss DWORD PTR [eax+4], xmm3
-
-		add coef_ptr, 16
-		add hist1_ptr, 8
-
-		mov eax, coef_ptr
-		movups xmm2, [eax]
-
-		mov eax, hist1_ptr
-		movlps xmm3, [eax]
-		shufps xmm3, xmm3, 44h
-		// hist1_ptr+1, hist1_ptr, hist1_ptr+1, hist1_ptr
-
-		mulps xmm2, xmm3
-
-		subss xmm1, xmm2
-		// Rotate elements right
-		shufps xmm2, xmm2, 39h
-		subss xmm1, xmm2
-
-		// Store new_hist
-		movss DWORD PTR [eax], xmm1
-
-		// Rotate elements right
-		shufps xmm2, xmm2, 39h
-		addss xmm1, xmm2
-
-		// Rotate elements right
-		shufps xmm2, xmm2, 39h
-		addss xmm1, xmm2
-
-		// Store previous hist
-		movss DWORD PTR [eax+4], xmm3
-
-		movss output, xmm1
-	}
-#else
-	output = atti386_iir_filter_sse(&output, hist1_ptr, coef_ptr);
-#endif
-	return output;
-}
-
-float iir_filter_3dnow(float input,float *hist1_ptr, float *coef_ptr) {
-	float output;
-
-	// 1st number of coefficients array is overall input scale factor, or filter gain
-	output = input * (*coef_ptr++);
-
-	// I find it very sad that 3DNow requires twice as many instructions as Intel's SSE
-	// Intel does have the upper hand here.
-#ifdef _MSC_VER
-	float tmp;
-	__asm {
-		movq mm1, output
-		mov ebx, coef_ptr
-		movq mm2, [ebx]
-
-		mov eax, hist1_ptr;
-		movq mm3, [eax]
-
-		pfmul mm2, mm3
-		pfsub mm1, mm2
-
-		psrlq mm2, 32
-		pfsub mm1, mm2
-
-		// Store new hist
-		movd tmp, mm1
-
-		add ebx, 8
-		movq mm2, [ebx]
-		movq mm3, [eax]
-
-		pfmul mm2, mm3
-		pfadd mm1, mm2
-
-		psrlq mm2, 32
-		pfadd mm1, mm2
-
-		push tmp
-		pop DWORD PTR [eax]
-
-		movd DWORD PTR [eax+4], mm3
-
-		add ebx, 8
-		add eax, 8
-
-		movq mm2, [ebx]
-		movq mm3, [eax]
-
-		pfmul mm2, mm3
-		pfsub mm1, mm2
-
-		psrlq mm2, 32
-		pfsub mm1, mm2
-
-		// Store new hist
-		movd tmp, mm1
-
-		add ebx, 8
-		movq mm2, [ebx]
-		movq mm3, [eax]
-
-		pfmul mm2, mm3
-		pfadd mm1, mm2
-
-		psrlq mm2, 32
-		pfadd mm1, mm2
-
-		push tmp
-		pop DWORD PTR [eax]
-		movd DWORD PTR [eax+4], mm3
-
-		movd output, mm1
-
-		femms
-	}
-#else
-	output = atti386_iir_filter_3DNow(output, hist1_ptr, coef_ptr);
-#endif
-	return output;
-}
-
-#if MT32EMU_USE_MMX > 0
-
-int i386_partialProductOutput(int len, Bit16s leftvol, Bit16s rightvol, Bit16s *partialBuf, Bit16s *mixedBuf) {
-	int tmplen = len >> 1;
-	if (tmplen == 0) {
-		return 0;
-	}
-#ifdef _MSC_VER
-	__asm {
-		mov ecx,tmplen
-		mov ax, leftvol
-		shl eax,16
-		mov ax, rightvol
-		movd mm1, eax
-		movd mm2, eax
-		psllq mm1, 32
-		por mm1, mm2
-		mov edi, partialBuf
-		mov esi, mixedBuf
-mmxloop1:
-		mov bx, [esi]
-		add esi,2
-		mov dx, [esi]
-		add esi,2
-
-		mov ax, dx
-		shl eax, 16
-		mov ax, dx
-		movd mm2,eax
-		psllq mm2, 32
-		mov ax, bx
-		shl eax, 16
-		mov ax, bx
-		movd mm3,eax
-		por mm2,mm3
-
-		pmulhw mm2, mm1
-		movq [edi], mm2
-		add edi, 8
-
-		dec ecx
-		cmp ecx,0
-		jg mmxloop1
-		emms
-	}
-#else
-	atti386_partialProductOutput(tmplen, leftvol, rightvol, partialBuf, mixedBuf);
-#endif
-	return tmplen << 1;
-}
-
-int i386_mixBuffers(Bit16s * buf1, Bit16s *buf2, int len) {
-	int tmplen = len >> 2;
-	if (tmplen == 0) {
-		return 0;
-	}
-#ifdef _MSC_VER
-	__asm {
-		mov ecx, tmplen
-		mov esi, buf1
-		mov edi, buf2
-
-mixloop1:
-		movq mm1, [edi]
-		movq mm2, [esi]
-		paddw mm1,mm2
-		movq [esi],mm1
-		add edi,8
-		add esi,8
-
-		dec ecx
-		cmp ecx,0
-		jg mixloop1
-		emms
-	}
-#else
-	atti386_mixBuffers(buf1, buf2, tmplen);
-#endif
-	return tmplen << 2;
-}
-
-
-int i386_mixBuffersRingMix(Bit16s * buf1, Bit16s *buf2, int len) {
-	int tmplen = len >> 2;
-	if (tmplen == 0) {
-		return 0;
-	}
-#ifdef _MSC_VER
-	__asm {
-		mov ecx, tmplen
-		mov esi, buf1
-		mov edi, buf2
-
-mixloop2:
-		movq mm1, [esi]
-		movq mm2, [edi]
-		movq mm3, mm1
-		pmulhw mm1, mm2
-		paddw mm1,mm3
-		movq [esi],mm1
-		add edi,8
-		add esi,8
-
-		dec ecx
-		cmp ecx,0
-		jg mixloop2
-		emms
-	}
-#else
-	atti386_mixBuffersRingMix(buf1, buf2, tmplen);
-#endif
-	return tmplen << 2;
-}
-
-int i386_mixBuffersRing(Bit16s * buf1, Bit16s *buf2, int len) {
-	int tmplen = len >> 2;
-	if (tmplen == 0) {
-		return 0;
-	}
-#ifdef _MSC_VER
-	__asm {
-		mov ecx, tmplen
-		mov esi, buf1
-		mov edi, buf2
-
-mixloop3:
-		movq mm1, [esi]
-		movq mm2, [edi]
-		pmulhw mm1, mm2
-		movq [esi],mm1
-		add edi,8
-		add esi,8
-
-		dec ecx
-		cmp ecx,0
-		jg mixloop3
-		emms
-	}
-#else
-	atti386_mixBuffersRing(buf1, buf2, tmplen);
-#endif
-	return tmplen << 2;
-}
-
-int i386_produceOutput1(Bit16s *useBuf, Bit16s *stream, Bit32u len, Bit16s volume) {
-	int tmplen = (len >> 1);
-	if (tmplen == 0) {
-		return 0;
-	}
-#ifdef _MSC_VER
-	__asm {
-		mov ecx, tmplen
-		mov ax,volume
-		shl eax,16
-		mov ax,volume
-		movd mm3,eax
-		movd mm2,eax
-		psllq mm3, 32
-		por mm3,mm2
-		mov esi, useBuf
-		mov edi, stream
-mixloop4:
-		movq mm1, [esi]
-		movq mm2, [edi]
-		pmulhw mm1, mm3
-		paddw mm1,mm2
-		movq [edi], mm1
-
-		add esi,8
-		add edi,8
-
-		dec ecx
-		cmp ecx,0
-		jg mixloop4
-		emms
-	}
-#else
-	atti386_produceOutput1(tmplen, volume, useBuf, stream);
-#endif
-	return tmplen << 1;
-}
-
-#endif
-
-}
-
-#endif
diff --git a/sound/softsynth/mt32/i386.h b/sound/softsynth/mt32/i386.h
deleted file mode 100644
index e8644411cd..0000000000
--- a/sound/softsynth/mt32/i386.h
+++ /dev/null
@@ -1,49 +0,0 @@
-/* Copyright (c) 2003-2005 Various contributors
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to
- * deal in the Software without restriction, including without limitation the
- * rights to use, copy, modify, merge, publish, distribute, sublicense, and/or
- * sell copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
- * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
- * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS
- * IN THE SOFTWARE.
- */
-
-#ifndef MT32EMU_I386_H
-#define MT32EMU_I386_H
-
-namespace MT32Emu {
-#ifdef MT32EMU_HAVE_X86
-
-// Function that detects the availablity of SSE SIMD instructions
-bool DetectSIMD();
-// Function that detects the availablity of 3DNow instructions
-bool Detect3DNow();
-
-float iir_filter_sse(float input,float *hist1_ptr, float *coef_ptr);
-float iir_filter_3dnow(float input,float *hist1_ptr, float *coef_ptr);
-float iir_filter_normal(float input,float *hist1_ptr, float *coef_ptr);
-
-#if MT32EMU_USE_MMX > 0
-int i386_partialProductOutput(int len, Bit16s leftvol, Bit16s rightvol, Bit16s *partialBuf, Bit16s *mixedBuf);
-int i386_mixBuffers(Bit16s * buf1, Bit16s *buf2, int len);
-int i386_mixBuffersRingMix(Bit16s * buf1, Bit16s *buf2, int len);
-int i386_mixBuffersRing(Bit16s * buf1, Bit16s *buf2, int len);
-int i386_produceOutput1(Bit16s *useBuf, Bit16s *stream, Bit32u len, Bit16s volume);
-#endif
-
-#endif
-
-}
-
-#endif
diff --git a/sound/softsynth/mt32/module.mk b/sound/softsynth/mt32/module.mk
deleted file mode 100644
index 4d5d899ac3..0000000000
--- a/sound/softsynth/mt32/module.mk
+++ /dev/null
@@ -1,14 +0,0 @@
-MODULE := sound/softsynth/mt32
-
-MODULE_OBJS := \
-	mt32_file.o \
-	i386.o \
-	part.o \
-	partial.o \
-	partialManager.o \
-	synth.o \
-	tables.o \
-	freeverb.o
-
-# Include common rules
-include $(srcdir)/rules.mk
diff --git a/sound/softsynth/mt32/mt32_file.cpp b/sound/softsynth/mt32/mt32_file.cpp
deleted file mode 100644
index cdf9fa13f6..0000000000
--- a/sound/softsynth/mt32/mt32_file.cpp
+++ /dev/null
@@ -1,70 +0,0 @@
-/* Copyright (c) 2003-2005 Various contributors
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to
- * deal in the Software without restriction, including without limitation the
- * rights to use, copy, modify, merge, publish, distribute, sublicense, and/or
- * sell copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
- * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
- * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS
- * IN THE SOFTWARE.
- */
-
-
-#include "mt32emu.h"
-
-namespace MT32Emu {
-
-bool File::readBit16u(Bit16u *in) {
-	Bit8u b[2];
-	if (read(&b[0], 2) != 2)
-		return false;
-	*in = ((b[0] << 8) | b[1]);
-	return true;
-}
-
-bool File::readBit32u(Bit32u *in) {
-	Bit8u b[4];
-	if (read(&b[0], 4) != 4)
-		return false;
-	*in = ((b[0] << 24) | (b[1] << 16) | (b[2] << 8) | b[3]);
-	return true;
-}
-
-bool File::writeBit16u(Bit16u out) {
-	if (!writeBit8u((Bit8u)((out & 0xFF00) >> 8))) {
-		return false;
-	}
-	if (!writeBit8u((Bit8u)(out & 0x00FF))) {
-		return false;
-	}
-	return true;
-}
-
-bool File::writeBit32u(Bit32u out) {
-	if (!writeBit8u((Bit8u)((out & 0xFF000000) >> 24))) {
-		return false;
-	}
-	if (!writeBit8u((Bit8u)((out & 0x00FF0000) >> 16))) {
-		return false;
-	}
-	if (!writeBit8u((Bit8u)((out & 0x0000FF00) >> 8))) {
-		return false;
-	}
-	if (!writeBit8u((Bit8u)(out & 0x000000FF))) {
-		return false;
-	}
-	return true;
-}
-
-} // End of namespace MT32Emu
-
diff --git a/sound/softsynth/mt32/mt32_file.h b/sound/softsynth/mt32/mt32_file.h
deleted file mode 100644
index e6641660ee..0000000000
--- a/sound/softsynth/mt32/mt32_file.h
+++ /dev/null
@@ -1,52 +0,0 @@
-/* Copyright (c) 2003-2005 Various contributors
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to
- * deal in the Software without restriction, including without limitation the
- * rights to use, copy, modify, merge, publish, distribute, sublicense, and/or
- * sell copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
- * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
- * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS
- * IN THE SOFTWARE.
- */
-
-#ifndef MT32EMU_FILE_H
-#define MT32EMU_FILE_H
-
-#include "common/scummsys.h"
-
-namespace MT32Emu {
-
-class File {
-public:
-	enum OpenMode {
-		OpenMode_read  = 0,
-		OpenMode_write = 1
-	};
-	virtual ~File() {}
-	virtual void close() = 0;
-	virtual size_t read(void *in, size_t size) = 0;
-	virtual bool readBit8u(Bit8u *in) = 0;
-	virtual bool readBit16u(Bit16u *in);
-	virtual bool readBit32u(Bit32u *in);
-	virtual size_t write(const void *out, size_t size) = 0;
-	virtual bool writeBit8u(Bit8u out) = 0;
-	// Note: May write a single byte to the file before failing
-	virtual bool writeBit16u(Bit16u out);
-	// Note: May write some (<4) bytes to the file before failing
-	virtual bool writeBit32u(Bit32u out);
-	virtual bool isEOF() = 0;
-};
-
-} // End of namespace MT32Emu
-
-#endif
diff --git a/sound/softsynth/mt32/mt32emu.h b/sound/softsynth/mt32/mt32emu.h
deleted file mode 100644
index 6eedf04bc0..0000000000
--- a/sound/softsynth/mt32/mt32emu.h
+++ /dev/null
@@ -1,70 +0,0 @@
-/* Copyright (c) 2003-2005 Various contributors
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to
- * deal in the Software without restriction, including without limitation the
- * rights to use, copy, modify, merge, publish, distribute, sublicense, and/or
- * sell copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
- * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
- * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS
- * IN THE SOFTWARE.
- */
-
-#ifndef MT32EMU_MT32EMU_H
-#define MT32EMU_MT32EMU_H
-
-// Debugging
-// Show the instruments played
-#define MT32EMU_MONITOR_INSTRUMENTS 1
-// Shows number of partials MT-32 is playing, and on which parts
-#define MT32EMU_MONITOR_PARTIALS 0
-// Determines how the waveform cache file is handled (must be regenerated after sampling rate change)
-#define MT32EMU_WAVECACHEMODE 0 // Load existing cache if possible, otherwise generate and save cache
-//#define MT32EMU_WAVECACHEMODE 1 // Load existing cache if possible, otherwise generate but don't save cache
-//#define MT32EMU_WAVECACHEMODE 2 // Ignore existing cache, generate and save cache
-//#define MT32EMU_WAVECACHEMODE 3 // Ignore existing cache, generate but don't save cache
-
-// Configuration
-// The maximum number of partials playing simultaneously
-#define MT32EMU_MAX_PARTIALS 32
-// The maximum number of notes playing simultaneously per part.
-// No point making it more than MT32EMU_MAX_PARTIALS, since each note needs at least one partial.
-#define MT32EMU_MAX_POLY 32
-// This calculates the exact frequencies of notes as they are played, instead of offsetting from pre-cached semitones. Potentially very slow.
-#define MT32EMU_ACCURATENOTES 0
-
-#if (defined (_MSC_VER) && defined(_M_IX86))
-#define MT32EMU_HAVE_X86
-#elif  defined(__GNUC__)
-#if __GNUC__ >= 3 && defined(__i386__)
-#define MT32EMU_HAVE_X86
-#endif
-#endif
-
-#ifdef MT32EMU_HAVE_X86
-#define MT32EMU_USE_MMX 1
-#else
-#define MT32EMU_USE_MMX 0
-#endif
-
-#include "freeverb.h"
-
-#include "structures.h"
-#include "i386.h"
-#include "mt32_file.h"
-#include "tables.h"
-#include "partial.h"
-#include "partialManager.h"
-#include "part.h"
-#include "synth.h"
-
-#endif
diff --git a/sound/softsynth/mt32/part.cpp b/sound/softsynth/mt32/part.cpp
deleted file mode 100644
index eb087f7ea0..0000000000
--- a/sound/softsynth/mt32/part.cpp
+++ /dev/null
@@ -1,633 +0,0 @@
-/* Copyright (c) 2003-2005 Various contributors
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to
- * deal in the Software without restriction, including without limitation the
- * rights to use, copy, modify, merge, publish, distribute, sublicense, and/or
- * sell copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
- * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
- * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS
- * IN THE SOFTWARE.
- */
-
-#include <string.h>
-#include <math.h>
-
-#include "mt32emu.h"
-
-namespace MT32Emu {
-
-static const Bit8u PartialStruct[13] = {
-	0, 0, 2, 2, 1, 3,
-	3, 0, 3, 0, 2, 1, 3 };
-
-static const Bit8u PartialMixStruct[13] = {
-	0, 1, 0, 1, 1, 0,
-	1, 3, 3, 2, 2, 2, 2 };
-
-static const float floatKeyfollow[17] = {
-	-1.0f, -1.0f/2.0f, -1.0f/4.0f, 0.0f,
-	1.0f/8.0f, 1.0f/4.0f, 3.0f/8.0f, 1.0f/2.0f, 5.0f/8.0f, 3.0f/4.0f, 7.0f/8.0f, 1.0f,
-	5.0f/4.0f, 3.0f/2.0f, 2.0f,
-	1.0009765625f, 1.0048828125f
-};
-
-//FIXME:KG: Put this dpoly stuff somewhere better
-bool dpoly::isActive() const {
-	return partials[0] != NULL || partials[1] != NULL || partials[2] != NULL || partials[3] != NULL;
-}
-
-Bit32u dpoly::getAge() const {
-	for (int i = 0; i < 4; i++) {
-		if (partials[i] != NULL) {
-			return partials[i]->age;
-		}
-	}
-	return 0;
-}
-
-RhythmPart::RhythmPart(Synth *useSynth, unsigned int usePartNum): Part(useSynth, usePartNum) {
-	strcpy(name, "Rhythm");
-	rhythmTemp = &synth->mt32ram.rhythmSettings[0];
-	refresh();
-}
-
-Part::Part(Synth *useSynth, unsigned int usePartNum) {
-	this->synth = useSynth;
-	this->partNum = usePartNum;
-	patchCache[0].dirty = true;
-	holdpedal = false;
-	patchTemp = &synth->mt32ram.patchSettings[partNum];
-	if (usePartNum == 8) {
-		// Nasty hack for rhythm
-		timbreTemp = NULL;
-	} else {
-		sprintf(name, "Part %d", partNum + 1);
-		timbreTemp = &synth->mt32ram.timbreSettings[partNum];
-	}
-	currentInstr[0] = 0;
-	currentInstr[10] = 0;
-	expression = 127;
-	volumeMult = 0;
-	volumesetting.leftvol = 32767;
-	volumesetting.rightvol = 32767;
-	bend = 0.0f;
-	memset(polyTable,0,sizeof(polyTable));
-	memset(patchCache, 0, sizeof(patchCache));
-}
-
-void Part::setHoldPedal(bool pedalval) {
-	if (holdpedal && !pedalval) {
-		holdpedal = false;
-		stopPedalHold();
-	} else {
-		holdpedal = pedalval;
-	}
-}
-
-void RhythmPart::setBend(unsigned int midiBend) {
-	synth->printDebug("%s: Setting bend (%d) not supported on rhythm", name, midiBend);
-	return;
-}
-
-void Part::setBend(unsigned int midiBend) {
-	// FIXME:KG: Slightly unbalanced increments, but I wanted min -1.0, centre 0.0 and max 1.0
-	if (midiBend <= 0x2000) {
-		bend = ((signed int)midiBend - 0x2000) / (float)0x2000;
-	} else {
-		bend = ((signed int)midiBend - 0x2000) / (float)0x1FFF;
-	}
-	// Loop through all partials to update their bend
-	for (int i = 0; i < MT32EMU_MAX_POLY; i++) {
-		for (int j = 0; j < 4; j++) {
-			if (polyTable[i].partials[j] != NULL) {
-				polyTable[i].partials[j]->setBend(bend);
-			}
-		}
-	}
-}
-
-void RhythmPart::setModulation(unsigned int midiModulation) {
-	synth->printDebug("%s: Setting modulation (%d) not supported on rhythm", name, midiModulation);
-}
-
-void Part::setModulation(unsigned int midiModulation) {
-	// Just a bloody guess, as always, before I get things figured out
-	for (int t = 0; t < 4; t++) {
-		if (patchCache[t].playPartial) {
-			int newrate = (patchCache[t].modsense * midiModulation) >> 7;
-			//patchCache[t].lfoperiod = lfotable[newrate];
-			patchCache[t].lfodepth = newrate;
-			//FIXME:KG: timbreTemp->partial[t].lfo.depth =
-		}
-	}
-}
-
-void RhythmPart::refresh() {
-	updateVolume();
-	// (Re-)cache all the mapped timbres ahead of time
-	for (unsigned int drumNum = 0; drumNum < synth->controlROMMap->rhythmSettingsCount; drumNum++) {
-		int drumTimbreNum = rhythmTemp[drumNum].timbre;
-		if (drumTimbreNum >= 127) // 94 on MT-32
-			continue;
-		Bit16s pan = rhythmTemp[drumNum].panpot; // They use R-L 0-14...
-		// FIXME:KG: Panning cache should be backed up to partials using it, too
-		if (pan < 7) {
-			drumPan[drumNum].leftvol = pan * 4681;
-			drumPan[drumNum].rightvol = 32767;
-		} else {
-			drumPan[drumNum].rightvol = (14 - pan) * 4681;
-			drumPan[drumNum].leftvol = 32767;
-		}
-		PatchCache *cache = drumCache[drumNum];
-		backupCacheToPartials(cache);
-		for (int t = 0; t < 4; t++) {
-			// Common parameters, stored redundantly
-			cache[t].dirty = true;
-			cache[t].pitchShift = 0.0f;
-			cache[t].benderRange = 0.0f;
-			cache[t].pansetptr = &drumPan[drumNum];
-			cache[t].reverb = rhythmTemp[drumNum].reverbSwitch > 0;
-		}
-	}
-}
-
-void Part::refresh() {
-	updateVolume();
-	backupCacheToPartials(patchCache);
-	for (int t = 0; t < 4; t++) {
-		// Common parameters, stored redundantly
-		patchCache[t].dirty = true;
-		patchCache[t].pitchShift = (patchTemp->patch.keyShift - 24) + (patchTemp->patch.fineTune - 50) / 100.0f;
-		patchCache[t].benderRange = patchTemp->patch.benderRange;
-		patchCache[t].pansetptr = &volumesetting;
-		patchCache[t].reverb = patchTemp->patch.reverbSwitch > 0;
-	}
-	memcpy(currentInstr, timbreTemp->common.name, 10);
-}
-
-const char *Part::getCurrentInstr() const {
-	return &currentInstr[0];
-}
-
-void RhythmPart::refreshTimbre(unsigned int absTimbreNum) {
-	for (int m = 0; m < 85; m++) {
-		if (rhythmTemp[m].timbre == absTimbreNum - 128)
-			drumCache[m][0].dirty = true;
-	}
-}
-
-void Part::refreshTimbre(unsigned int absTimbreNum) {
-	if (getAbsTimbreNum() == absTimbreNum) {
-		memcpy(currentInstr, timbreTemp->common.name, 10);
-		patchCache[0].dirty = true;
-	}
-}
-
-int Part::fixBiaslevel(int srcpnt, int *dir) {
-	int noteat = srcpnt & 0x3F;
-	int outnote;
-	if (srcpnt < 64)
-		*dir = 0;
-	else
-		*dir = 1;
-	outnote = 33 + noteat;
-	//synth->printDebug("Bias note %d, dir %d", outnote, *dir);
-
-	return outnote;
-}
-
-int Part::fixKeyfollow(int srckey) {
-	if (srckey>=0 && srckey<=16) {
-		int keyfix[17] = { -256*16, -128*16, -64*16, 0, 32*16, 64*16, 96*16, 128*16, (128+32)*16, 192*16, (192+32)*16, 256*16, (256+64)*16, (256+128)*16, (512)*16, 4100, 4116};
-		return keyfix[srckey];
-	} else {
-		//LOG(LOG_ERROR|LOG_MISC,"Missed key: %d", srckey);
-		return 256;
-	}
-}
-
-void Part::abortPoly(dpoly *poly) {
-	if (!poly->isPlaying) {
-		return;
-	}
-	for (int i = 0; i < 4; i++) {
-		Partial *partial = poly->partials[i];
-		if (partial != NULL) {
-			partial->deactivate();
-		}
-	}
-	poly->isPlaying = false;
-}
-
-void Part::setPatch(const PatchParam *patch) {
-	patchTemp->patch = *patch;
-}
-
-void RhythmPart::setTimbre(TimbreParam * /*timbre*/) {
-	synth->printDebug("%s: Attempted to call setTimbre() - doesn't make sense for rhythm", name);
-}
-
-void Part::setTimbre(TimbreParam *timbre) {
-	*timbreTemp = *timbre;
-}
-
-unsigned int RhythmPart::getAbsTimbreNum() const {
-	synth->printDebug("%s: Attempted to call getAbsTimbreNum() - doesn't make sense for rhythm", name);
-	return 0;
-}
-
-unsigned int Part::getAbsTimbreNum() const {
-	return (patchTemp->patch.timbreGroup * 64) + patchTemp->patch.timbreNum;
-}
-
-void RhythmPart::setProgram(unsigned int patchNum) {
-	synth->printDebug("%s: Attempt to set program (%d) on rhythm is invalid", name, patchNum);
-}
-
-void Part::setProgram(unsigned int patchNum) {
-	setPatch(&synth->mt32ram.patches[patchNum]);
-	setTimbre(&synth->mt32ram.timbres[getAbsTimbreNum()].timbre);
-
-	refresh();
-
-	allSoundOff(); //FIXME:KG: Is this correct?
-}
-
-void Part::backupCacheToPartials(PatchCache cache[4]) {
-	// check if any partials are still playing with the old patch cache
-	// if so then duplicate the cached data from the part to the partial so that
-	// we can change the part's cache without affecting the partial.
-	// We delay this until now to avoid a copy operation with every note played
-	for (int m = 0; m < MT32EMU_MAX_POLY; m++) {
-		for (int i = 0; i < 4; i++) {
-			Partial *partial = polyTable[m].partials[i];
-			if (partial != NULL && partial->patchCache == &cache[i]) {
-				partial->cachebackup = cache[i];
-				partial->patchCache = &partial->cachebackup;
-			}
-		}
-	}
-}
-
-void Part::cacheTimbre(PatchCache cache[4], const TimbreParam *timbre) {
-	backupCacheToPartials(cache);
-	int partialCount = 0;
-	for (int t = 0; t < 4; t++) {
-		cache[t].PCMPartial = false;
-		if (((timbre->common.pmute >> t) & 0x1) == 1) {
-			cache[t].playPartial = true;
-			partialCount++;
-		} else {
-			cache[t].playPartial = false;
-			continue;
-		}
-
-		// Calculate and cache common parameters
-
-		cache[t].pcm = timbre->partial[t].wg.pcmwave;
-		cache[t].useBender = (timbre->partial[t].wg.bender == 1);
-
-		switch (t) {
-		case 0:
-			cache[t].PCMPartial = (PartialStruct[(int)timbre->common.pstruct12] & 0x2) ? true : false;
-			cache[t].structureMix = PartialMixStruct[(int)timbre->common.pstruct12];
-			cache[t].structurePosition = 0;
-			cache[t].structurePair = 1;
-			break;
-		case 1:
-			cache[t].PCMPartial = (PartialStruct[(int)timbre->common.pstruct12] & 0x1) ? true : false;
-			cache[t].structureMix = PartialMixStruct[(int)timbre->common.pstruct12];
-			cache[t].structurePosition = 1;
-			cache[t].structurePair = 0;
-			break;
-		case 2:
-			cache[t].PCMPartial = (PartialStruct[(int)timbre->common.pstruct34] & 0x2) ? true : false;
-			cache[t].structureMix = PartialMixStruct[(int)timbre->common.pstruct34];
-			cache[t].structurePosition = 0;
-			cache[t].structurePair = 3;
-			break;
-		case 3:
-			cache[t].PCMPartial = (PartialStruct[(int)timbre->common.pstruct34] & 0x1) ? true : false;
-			cache[t].structureMix = PartialMixStruct[(int)timbre->common.pstruct34];
-			cache[t].structurePosition = 1;
-			cache[t].structurePair = 2;
-			break;
-		default:
-			break;
-		}
-
-		cache[t].waveform = timbre->partial[t].wg.waveform;
-		cache[t].pulsewidth = timbre->partial[t].wg.pulsewid;
-		cache[t].pwsens = timbre->partial[t].wg.pwvelo;
-		if (timbre->partial[t].wg.keyfollow > 16) {
-			synth->printDebug("Bad keyfollow value in timbre!");
-			cache[t].pitchKeyfollow = 1.0f;
-		} else {
-			cache[t].pitchKeyfollow = floatKeyfollow[timbre->partial[t].wg.keyfollow];
-		}
-
-		cache[t].pitch = timbre->partial[t].wg.coarse + (timbre->partial[t].wg.fine - 50) / 100.0f + 24.0f;
-		cache[t].pitchEnv = timbre->partial[t].env;
-		cache[t].pitchEnv.sensitivity = (char)((float)cache[t].pitchEnv.sensitivity * 1.27f);
-		cache[t].pitchsustain = cache[t].pitchEnv.level[3];
-
-		// Calculate and cache TVA envelope stuff
-		cache[t].ampEnv = timbre->partial[t].tva;
-		cache[t].ampEnv.level = (char)((float)cache[t].ampEnv.level * 1.27f);
-
-		cache[t].ampbias[0] = fixBiaslevel(cache[t].ampEnv.biaspoint1, &cache[t].ampdir[0]);
-		cache[t].ampblevel[0] = 12 - cache[t].ampEnv.biaslevel1;
-		cache[t].ampbias[1] = fixBiaslevel(cache[t].ampEnv.biaspoint2, &cache[t].ampdir[1]);
-		cache[t].ampblevel[1] = 12 - cache[t].ampEnv.biaslevel2;
-		cache[t].ampdepth = cache[t].ampEnv.envvkf * cache[t].ampEnv.envvkf;
-
-		// Calculate and cache filter stuff
-		cache[t].filtEnv = timbre->partial[t].tvf;
-		cache[t].filtkeyfollow  = fixKeyfollow(cache[t].filtEnv.keyfollow);
-		cache[t].filtEnv.envdepth = (char)((float)cache[t].filtEnv.envdepth * 1.27);
-		cache[t].tvfbias = fixBiaslevel(cache[t].filtEnv.biaspoint, &cache[t].tvfdir);
-		cache[t].tvfblevel = cache[t].filtEnv.biaslevel;
-		cache[t].filtsustain  = cache[t].filtEnv.envlevel[3];
-
-		// Calculate and cache LFO stuff
-		cache[t].lfodepth = timbre->partial[t].lfo.depth;
-		cache[t].lfoperiod = synth->tables.lfoPeriod[(int)timbre->partial[t].lfo.rate];
-		cache[t].lforate = timbre->partial[t].lfo.rate;
-		cache[t].modsense = timbre->partial[t].lfo.modsense;
-	}
-	for (int t = 0; t < 4; t++) {
-		// Common parameters, stored redundantly
-		cache[t].dirty = false;
-		cache[t].partialCount = partialCount;
-		cache[t].sustain = (timbre->common.nosustain == 0);
-	}
-	//synth->printDebug("Res 1: %d 2: %d 3: %d 4: %d", cache[0].waveform, cache[1].waveform, cache[2].waveform, cache[3].waveform);
-
-#if MT32EMU_MONITOR_INSTRUMENTS == 1
-	synth->printDebug("%s (%s): Recached timbre", name, currentInstr);
-	for (int i = 0; i < 4; i++) {
-		synth->printDebug(" %d: play=%s, pcm=%s (%d), wave=%d", i, cache[i].playPartial ? "YES" : "NO", cache[i].PCMPartial ? "YES" : "NO", timbre->partial[i].wg.pcmwave, timbre->partial[i].wg.waveform);
-	}
-#endif
-}
-
-const char *Part::getName() const {
-	return name;
-}
-
-void Part::updateVolume() {
-	volumeMult = synth->tables.volumeMult[patchTemp->outlevel * expression / 127];
-}
-
-int Part::getVolume() const {
-	// FIXME: Use the mappings for this in the control ROM
-	return patchTemp->outlevel * 127 / 100;
-}
-
-void Part::setVolume(int midiVolume) {
-	// FIXME: Use the mappings for this in the control ROM
-	patchTemp->outlevel = (Bit8u)(midiVolume * 100 / 127);
-	updateVolume();
-	synth->printDebug("%s (%s): Set volume to %d", name, currentInstr, midiVolume);
-}
-
-void Part::setExpression(int midiExpression) {
-	expression = midiExpression;
-	updateVolume();
-}
-
-void RhythmPart::setPan(unsigned int midiPan)
-{
-	// FIXME:KG: This is unchangeable for drums (they always use drumPan), is that correct?
-	synth->printDebug("%s: Setting pan (%d) not supported on rhythm", name, midiPan);
-}
-
-void Part::setPan(unsigned int midiPan) {
-	// FIXME:KG: Tweaked this a bit so that we have a left 100%, centre and right 100%
-	// (But this makes the range somewhat skewed)
-	// Check against the real thing
-	// NOTE: Panning is inverted compared to GM.
-	if (midiPan < 64) {
-		volumesetting.leftvol = (Bit16s)(midiPan * 512);
-		volumesetting.rightvol = 32767;
-	} else if (midiPan == 64) {
-		volumesetting.leftvol = 32767;
-		volumesetting.rightvol = 32767;
-	} else {
-		volumesetting.rightvol = (Bit16s)((127 - midiPan) * 520);
-		volumesetting.leftvol = 32767;
-	}
-	patchTemp->panpot = (Bit8u)(midiPan * 14 / 127);
-	//synth->printDebug("%s (%s): Set pan to %d", name, currentInstr, panpot);
-}
-
-void RhythmPart::playNote(unsigned int key, int vel) {
-	if (key < 24 || key > 108)/*> 87 on MT-32)*/ {
-		synth->printDebug("%s: Attempted to play invalid key %d", name, key);
-		return;
-	}
-	int drumNum = key - 24;
-	int drumTimbreNum = rhythmTemp[drumNum].timbre;
-	if (drumTimbreNum >= 127) { // 94 on MT-32
-		synth->printDebug("%s: Attempted to play unmapped key %d", name, key);
-		return;
-	}
-	int absTimbreNum = drumTimbreNum + 128;
-	TimbreParam *timbre = &synth->mt32ram.timbres[absTimbreNum].timbre;
-	memcpy(currentInstr, timbre->common.name, 10);
-#if MT32EMU_MONITOR_INSTRUMENTS == 1
-	synth->printDebug("%s (%s): starting poly (drum %d, timbre %d) - Vel %d Key %d", name, currentInstr, drumNum, absTimbreNum, vel, key);
-#endif
-	if (drumCache[drumNum][0].dirty) {
-		cacheTimbre(drumCache[drumNum], timbre);
-	}
-	playPoly(drumCache[drumNum], key, MIDDLEC, vel);
-}
-
-void Part::playNote(unsigned int key, int vel) {
-	int freqNum = key;
-	if (freqNum < 12) {
-		synth->printDebug("%s (%s): Attempted to play invalid key %d < 12; moving up by octave", name, currentInstr, key);
-		freqNum += 12;
-	} else if (freqNum > 108) {
-		synth->printDebug("%s (%s): Attempted to play invalid key %d > 108; moving down by octave", name, currentInstr, key);
-		while (freqNum > 108) {
-			freqNum -= 12;
-		}
-	}
-	// POLY1 mode, Single Assign
-	// Haven't found any software that uses any of the other poly modes
-	// FIXME:KG: Should this also apply to rhythm?
-	for (unsigned int i = 0; i < MT32EMU_MAX_POLY; i++) {
-		if (polyTable[i].isActive() && (polyTable[i].key == key)) {
-			//AbortPoly(&polyTable[i]);
-			stopNote(key);
-			break;
-		}
-	}
-#if MT32EMU_MONITOR_INSTRUMENTS == 1
-	synth->printDebug("%s (%s): starting poly - Vel %d Key %d", name, currentInstr, vel, key);
-#endif
-	if (patchCache[0].dirty) {
-		cacheTimbre(patchCache, timbreTemp);
-	}
-	playPoly(patchCache, key, freqNum, vel);
-}
-
-void Part::playPoly(const PatchCache cache[4], unsigned int key, int freqNum, int vel) {
-	unsigned int needPartials = cache[0].partialCount;
-	unsigned int freePartials = synth->partialManager->getFreePartialCount();
-
-	if (freePartials < needPartials) {
-		if (!synth->partialManager->freePartials(needPartials - freePartials, partNum)) {
-			synth->printDebug("%s (%s): Insufficient free partials to play key %d (vel=%d); needed=%d, free=%d", name, currentInstr, key, vel, needPartials, synth->partialManager->getFreePartialCount());
-			return;
-		}
-	}
-	// Find free poly
-	int m;
-	for (m = 0; m < MT32EMU_MAX_POLY; m++) {
-		if (!polyTable[m].isActive()) {
-			break;
-		}
-	}
-	if (m == MT32EMU_MAX_POLY) {
-		synth->printDebug("%s (%s): No free poly to play key %d (vel %d)", name, currentInstr, key, vel);
-		return;
-	}
-
-	dpoly *tpoly = &polyTable[m];
-
-	tpoly->isPlaying = true;
-	tpoly->key = key;
-	tpoly->isDecay = false;
-	tpoly->freqnum = freqNum;
-	tpoly->vel = vel;
-	tpoly->pedalhold = false;
-
-	bool allnull = true;
-	for (int x = 0; x < 4; x++) {
-		if (cache[x].playPartial) {
-			tpoly->partials[x] = synth->partialManager->allocPartial(partNum);
-			allnull = false;
-		} else {
-			tpoly->partials[x] = NULL;
-		}
-	}
-
-	if (allnull)
-		synth->printDebug("%s (%s): No partials to play for this instrument", name, this->currentInstr);
-
-	tpoly->sustain = cache[0].sustain;
-	tpoly->volumeptr = &volumeMult;
-
-	for (int x = 0; x < 4; x++) {
-		if (tpoly->partials[x] != NULL) {
-			tpoly->partials[x]->startPartial(tpoly, &cache[x], tpoly->partials[cache[x].structurePair]);
-			tpoly->partials[x]->setBend(bend);
-		}
-	}
-}
-
-static void startDecayPoly(dpoly *tpoly) {
-	if (tpoly->isDecay) {
-		return;
-	}
-	tpoly->isDecay = true;
-
-	for (int t = 0; t < 4; t++) {
-		Partial *partial = tpoly->partials[t];
-		if (partial == NULL)
-			continue;
-		partial->startDecayAll();
-	}
-	tpoly->isPlaying = false;
-}
-
-void Part::allNotesOff() {
-	// Note: Unchecked on real MT-32, but the MIDI specification states that all notes off (0x7B)
-	// should treat the hold pedal as usual.
-	// All *sound* off (0x78) should stop notes immediately regardless of the hold pedal.
-	// The latter controller is not implemented on the MT-32 (according to the docs).
-	for (int q = 0; q < MT32EMU_MAX_POLY; q++) {
-		dpoly *tpoly = &polyTable[q];
-		if (tpoly->isPlaying) {
-			if (holdpedal)
-				tpoly->pedalhold = true;
-			else if (tpoly->sustain)
-				startDecayPoly(tpoly);
-		}
-	}
-}
-
-void Part::allSoundOff() {
-	for (int q = 0; q < MT32EMU_MAX_POLY; q++) {
-		dpoly *tpoly = &polyTable[q];
-		if (tpoly->isPlaying) {
-			startDecayPoly(tpoly);
-		}
-	}
-}
-
-void Part::stopPedalHold() {
-	for (int q = 0; q < MT32EMU_MAX_POLY; q++) {
-		dpoly *tpoly;
-		tpoly = &polyTable[q];
-		if (tpoly->isActive() && tpoly->pedalhold)
-			stopNote(tpoly->key);
-	}
-}
-
-void Part::stopNote(unsigned int key) {
-	// Non-sustaining instruments ignore stop commands.
-	// They die away eventually anyway
-
-#if MT32EMU_MONITOR_INSTRUMENTS == 1
-	synth->printDebug("%s (%s): stopping key %d", name, currentInstr, key);
-#endif
-
-	if (key != 255) {
-		for (int q = 0; q < MT32EMU_MAX_POLY; q++) {
-			dpoly *tpoly = &polyTable[q];
-			if (tpoly->isPlaying && tpoly->key == key) {
-				if (holdpedal)
-					tpoly->pedalhold = true;
-				else if (tpoly->sustain)
-					startDecayPoly(tpoly);
-			}
-		}
-		return;
-	}
-
-	// Find oldest poly... yes, the MT-32 can be reconfigured to kill different poly first
-	// This is simplest
-	int oldest = -1;
-	Bit32u oldage = 0;
-
-	for (int q = 0; q < MT32EMU_MAX_POLY; q++) {
-		dpoly *tpoly = &polyTable[q];
-
-		if (tpoly->isPlaying && !tpoly->isDecay) {
-			if (tpoly->getAge() >= oldage) {
-				oldage = tpoly->getAge();
-				oldest = q;
-			}
-		}
-	}
-
-	if (oldest != -1) {
-		startDecayPoly(&polyTable[oldest]);
-	}
-}
-
-}
diff --git a/sound/softsynth/mt32/part.h b/sound/softsynth/mt32/part.h
deleted file mode 100644
index 54c4999653..0000000000
--- a/sound/softsynth/mt32/part.h
+++ /dev/null
@@ -1,113 +0,0 @@
-/* Copyright (c) 2003-2005 Various contributors
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to
- * deal in the Software without restriction, including without limitation the
- * rights to use, copy, modify, merge, publish, distribute, sublicense, and/or
- * sell copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
- * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
- * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS
- * IN THE SOFTWARE.
- */
-
-#ifndef MT32EMU_PART_H
-#define MT32EMU_PART_H
-
-namespace MT32Emu {
-
-class PartialManager;
-class Synth;
-
-class Part {
-private:
-	// Pointers to the areas of the MT-32's memory dedicated to this part (for parts 1-8)
-	MemParams::PatchTemp *patchTemp;
-	TimbreParam *timbreTemp;
-
-	// 0=Part 1, .. 7=Part 8, 8=Rhythm
-	unsigned int partNum;
-
-	bool holdpedal;
-
-	StereoVolume volumesetting;
-
-	PatchCache patchCache[4];
-
-	float bend; // -1.0 .. +1.0
-
-	dpoly polyTable[MT32EMU_MAX_POLY];
-
-	void abortPoly(dpoly *poly);
-
-	static int fixKeyfollow(int srckey);
-	static int fixBiaslevel(int srcpnt, int *dir);
-
-	void setPatch(const PatchParam *patch);
-
-protected:
-	Synth *synth;
-	char name[8]; // "Part 1".."Part 8", "Rhythm"
-	char currentInstr[11];
-	int expression;
-	Bit32u volumeMult;
-
-	void updateVolume();
-	void backupCacheToPartials(PatchCache cache[4]);
-	void cacheTimbre(PatchCache cache[4], const TimbreParam *timbre);
-	void playPoly(const PatchCache cache[4], unsigned int key, int freqNum, int vel);
-	const char *getName() const;
-
-public:
-	Part(Synth *synth, unsigned int usePartNum);
-	virtual ~Part() {}
-	virtual void playNote(unsigned int key, int vel);
-	void stopNote(unsigned int key);
-	void allNotesOff();
-	void allSoundOff();
-	int getVolume() const;
-	void setVolume(int midiVolume);
-	void setExpression(int midiExpression);
-	virtual void setPan(unsigned int midiPan);
-	virtual void setBend(unsigned int midiBend);
-	virtual void setModulation(unsigned int midiModulation);
-	virtual void setProgram(unsigned int midiProgram);
-	void setHoldPedal(bool pedalval);
-	void stopPedalHold();
-	virtual void refresh();
-	virtual void refreshTimbre(unsigned int absTimbreNum);
-	virtual void setTimbre(TimbreParam *timbre);
-	virtual unsigned int getAbsTimbreNum() const;
-	const char *getCurrentInstr() const;
-};
-
-class RhythmPart: public Part {
-	// Pointer to the area of the MT-32's memory dedicated to rhythm
-	const MemParams::RhythmTemp *rhythmTemp;
-
-	// This caches the timbres/settings in use by the rhythm part
-	PatchCache drumCache[85][4];
-	StereoVolume drumPan[85];
-public:
-	RhythmPart(Synth *synth, unsigned int usePartNum);
-	void refresh();
-	void refreshTimbre(unsigned int timbreNum);
-	void setTimbre(TimbreParam *timbre);
-	void playNote(unsigned int key, int vel);
-	unsigned int getAbsTimbreNum() const;
-	void setPan(unsigned int midiPan);
-	void setBend(unsigned int midiBend);
-	void setModulation(unsigned int midiModulation);
-	void setProgram(unsigned int patchNum);
-};
-
-}
-#endif
diff --git a/sound/softsynth/mt32/partial.cpp b/sound/softsynth/mt32/partial.cpp
deleted file mode 100644
index 5ba9ef6145..0000000000
--- a/sound/softsynth/mt32/partial.cpp
+++ /dev/null
@@ -1,968 +0,0 @@
-/* Copyright (c) 2003-2005 Various contributors
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to
- * deal in the Software without restriction, including without limitation the
- * rights to use, copy, modify, merge, publish, distribute, sublicense, and/or
- * sell copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
- * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
- * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS
- * IN THE SOFTWARE.
- */
-
-#include <stdlib.h>
-#include <math.h>
-#include <string.h>
-
-#include "mt32emu.h"
-
-#if defined(MACOSX) || defined(SOLARIS) || defined(__MINGW32__)
-// Older versions of Mac OS X didn't supply a powf function, so using it
-// will cause a binary incompatibility when trying to run a binary built
-// on a newer OS X release on an olderr one. And Solaris 8 doesn't provide
-// powf, floorf, fabsf etc. at all.
-// Cross-compiled MinGW32 toolchains suffer from a cross-compile bug in
-// libstdc++. math/stubs.o should be empty, but it comes with a symbol for
-// powf, resulting in a linker error because of multiple definitions.
-// Hence we re-define them here. The only potential drawback is that it
-// might be a little bit slower this way.
-#define powf(x,y)	((float)pow(x,y))
-#define floorf(x)	((float)floor(x))
-#define fabsf(x)	((float)fabs(x))
-#endif
-
-#define FIXEDPOINT_UDIV(x, y, point) (((x) << (point)) / ((y)))
-#define FIXEDPOINT_SDIV(x, y, point) (((x) * (1 << point)) / ((y)))
-#define FIXEDPOINT_UMULT(x, y, point) (((x) * (y)) >> point)
-#define FIXEDPOINT_SMULT(x, y, point) (((x) * (y)) / (1 << point))
-
-using namespace MT32Emu;
-
-Partial::Partial(Synth *useSynth) {
-	this->synth = useSynth;
-	ownerPart = -1;
-	poly = NULL;
-	pair = NULL;
-#if MT32EMU_ACCURATENOTES == 1
-	for (int i = 0; i < 3; i++) {
-		noteLookupStorage.waveforms[i] = new Bit16s[65536];
-	}
-	noteLookup = &noteLookupStorage;
-#endif
-}
-
-Partial::~Partial() {
-#if MT32EMU_ACCURATENOTES == 1
-	for (int i = 0; i < 3; i++) {
-		delete[] noteLookupStorage.waveforms[i];
-	}
-	delete[] noteLookupStorage.wavTable;
-#endif
-}
-
-int Partial::getOwnerPart() const {
-	return ownerPart;
-}
-
-bool Partial::isActive() {
-	return ownerPart > -1;
-}
-
-const dpoly *Partial::getDpoly() const {
-	return this->poly;
-}
-
-void Partial::activate(int part) {
-	// This just marks the partial as being assigned to a part
-	ownerPart = part;
-}
-
-void Partial::deactivate() {
-	ownerPart = -1;
-	if (poly != NULL) {
-		for (int i = 0; i < 4; i++) {
-			if (poly->partials[i] == this) {
-				poly->partials[i] = NULL;
-				break;
-			}
-		}
-		if (pair != NULL) {
-			pair->pair = NULL;
-		}
-	}
-}
-
-void Partial::initKeyFollow(int key) {
-	// Setup partial keyfollow
-	// Note follow relative to middle C
-
-	// Calculate keyfollow for pitch
-#if 1
-	float rel = key == -1 ? 0.0f : (key - MIDDLEC);
-	float newPitch = rel * patchCache->pitchKeyfollow + patchCache->pitch + patchCache->pitchShift;
-	//FIXME:KG: Does it truncate the keyfollowed pitch to a semitone (towards MIDDLEC)?
-	//int newKey = (int)(rel * patchCache->pitchKeyfollow);
-	//float newPitch = newKey + patchCache->pitch + patchCache->pitchShift;
-#else
-	float rel = key == -1 ? 0.0f : (key + patchCache->pitchShift - MIDDLEC);
-	float newPitch = rel * patchCache->pitchKeyfollow + patchCache->pitch;
-#endif
-#if MT32EMU_ACCURATENOTES == 1
-	noteVal = newPitch;
-	synth->printDebug("key=%d, pitch=%f, pitchKeyfollow=%f, pitchShift=%f, newPitch=%f", key, (double)patchCache->pitch, (double)patchCache->pitchKeyfollow, (double)patchCache->pitchShift, (double)newPitch);
-#else
-	float newPitchInt;
-	float newPitchFract = modff(newPitch, &newPitchInt);
-	if (newPitchFract > 0.5f) {
-		newPitchInt += 1.0f;
-		newPitchFract -= 1.0f;
-	}
-	noteVal = (int)newPitchInt;
-	fineShift = (int)(powf(2.0f, newPitchFract / 12.0f) * 4096.0f);
-	synth->printDebug("key=%d, pitch=%f, pitchKeyfollow=%f, pitchShift=%f, newPitch=%f, noteVal=%d, fineShift=%d", key, (double)patchCache->pitch, (double)patchCache->pitchKeyfollow, (double)patchCache->pitchShift, (double)newPitch, noteVal, fineShift);
-#endif
-	// FIXME:KG: Raise/lower by octaves until in the supported range.
-	while (noteVal > HIGHEST_NOTE) // FIXME:KG: see tables.cpp: >108?
-		noteVal -= 12;
-	while (noteVal < LOWEST_NOTE) // FIXME:KG: see tables.cpp: <12?
-		noteVal += 12;
-	// Calculate keyfollow for filter
-	int keyfollow = ((key - MIDDLEC) * patchCache->filtkeyfollow) / 4096;
-	if (keyfollow > 108)
-		keyfollow = 108;
-	else if (keyfollow < -108)
-		keyfollow = -108;
-	filtVal = synth->tables.tvfKeyfollowMult[keyfollow + 108];
-	realVal = synth->tables.tvfKeyfollowMult[(noteVal - MIDDLEC) + 108];
-}
-
-int Partial::getKey() const {
-	if (poly == NULL) {
-		return -1;
-	} else {
-		return poly->key;
-	}
-}
-
-void Partial::startPartial(dpoly *usePoly, const PatchCache *useCache, Partial *pairPartial) {
-	if (usePoly == NULL || useCache == NULL) {
-		synth->printDebug("*** Error: Starting partial for owner %d, usePoly=%s, useCache=%s", ownerPart, usePoly == NULL ? "*** NULL ***" : "OK", useCache == NULL ? "*** NULL ***" : "OK");
-		return;
-	}
-	patchCache = useCache;
-	poly = usePoly;
-	mixType = patchCache->structureMix;
-	structurePosition = patchCache->structurePosition;
-
-	play = true;
-	initKeyFollow(poly->freqnum); // Initialises noteVal, filtVal and realVal
-#if MT32EMU_ACCURATENOTES == 0
-	noteLookup = &synth->tables.noteLookups[noteVal - LOWEST_NOTE];
-#else
-	Tables::initNote(synth, &noteLookupStorage, noteVal, (float)synth->myProp.sampleRate, synth->masterTune, synth->pcmWaves, NULL);
-#endif
-	keyLookup = &synth->tables.keyLookups[poly->freqnum - 12];
-
-	if (patchCache->PCMPartial) {
-		pcmNum = patchCache->pcm;
-		if (synth->controlROMMap->pcmCount > 128) {
-			// CM-32L, etc. support two "banks" of PCMs, selectable by waveform type parameter.
-			if (patchCache->waveform > 1) {
-				pcmNum += 128;
-			}
-		}
-		pcmWave = &synth->pcmWaves[pcmNum];
-	} else {
-		pcmWave = NULL;
-	}
-
-	lfoPos = 0;
-	pulsewidth = patchCache->pulsewidth + synth->tables.pwVelfollowAdd[patchCache->pwsens][poly->vel];
-	if (pulsewidth > 100) {
-		pulsewidth = 100;
-	} else if (pulsewidth < 0) {
-		pulsewidth = 0;
-	}
-
-	for (int e = 0; e < 3; e++) {
-		envs[e].envpos = 0;
-		envs[e].envstat = -1;
-		envs[e].envbase = 0;
-		envs[e].envdist = 0;
-		envs[e].envsize = 0;
-		envs[e].sustaining = false;
-		envs[e].decaying = false;
-		envs[e].prevlevel = 0;
-		envs[e].counter = 0;
-		envs[e].count = 0;
-	}
-	ampEnvVal = 0;
-	pitchEnvVal = 0;
-	pitchSustain = false;
-	loopPos = 0;
-	partialOff.pcmoffset = partialOff.pcmplace = 0;
-	pair = pairPartial;
-	useNoisePair = pairPartial == NULL && (mixType == 1 || mixType == 2);
-	age = 0;
-	alreadyOutputed = false;
-	memset(history,0,sizeof(history));
-}
-
-Bit16s *Partial::generateSamples(long length) {
-	if (!isActive() || alreadyOutputed) {
-		return NULL;
-	}
-	if (poly == NULL) {
-		synth->printDebug("*** ERROR: poly is NULL at Partial::generateSamples()!");
-		return NULL;
-	}
-
-	alreadyOutputed = true;
-
-	// Generate samples
-
-	Bit16s *partialBuf = &myBuffer[0];
-	Bit32u volume = *poly->volumeptr;
-	while (length--) {
-		Bit32s envval;
-		Bit32s sample = 0;
-		if (!envs[EnvelopeType_amp].sustaining) {
-			if (envs[EnvelopeType_amp].count <= 0) {
-				Bit32u ampval = getAmpEnvelope();
-				if (!play) {
-					deactivate();
-					break;
-				}
-				if (ampval > 100) {
-					ampval = 100;
-				}
-
-				ampval = synth->tables.volumeMult[ampval];
-				ampval = FIXEDPOINT_UMULT(ampval, synth->tables.tvaVelfollowMult[poly->vel][(int)patchCache->ampEnv.velosens], 8);
-				//if (envs[EnvelopeType_amp].sustaining)
-				ampEnvVal = ampval;
-			}
-			--envs[EnvelopeType_amp].count;
-		}
-
-		unsigned int lfoShift = 0x1000;
-		if (pitchSustain) {
-			// Calculate LFO position
-			// LFO does not kick in completely until pitch envelope sustains
-			if (patchCache->lfodepth > 0) {
-				lfoPos++;
-				if (lfoPos >= patchCache->lfoperiod)
-					lfoPos = 0;
-				int lfoatm = FIXEDPOINT_UDIV(lfoPos, patchCache->lfoperiod, 16);
-				int lfoatr = synth->tables.sintable[lfoatm];
-				lfoShift = synth->tables.lfoShift[patchCache->lfodepth][lfoatr];
-			}
-		} else {
-			// Calculate Pitch envelope
-			envval = getPitchEnvelope();
-			int pd = patchCache->pitchEnv.depth;
-			pitchEnvVal = synth->tables.pitchEnvVal[pd][envval];
-		}
-
-		int delta;
-
-		// Wrap positions or end if necessary
-		if (patchCache->PCMPartial) {
-			// PCM partial
-
-			delta = noteLookup->wavTable[pcmNum];
-			int len = pcmWave->len;
-			if (partialOff.pcmplace >= len) {
-				if (pcmWave->loop) {
-					//partialOff.pcmplace = partialOff.pcmoffset = 0;
-					partialOff.pcmplace %= len;
-				} else {
-					play = false;
-					deactivate();
-					break;
-				}
-			}
-		} else {
-			// Synthesis partial
-			delta = 0x10000;
-			partialOff.pcmplace %= (Bit16u)noteLookup->div2;
-		}
-
-		// Build delta for position of next sample
-		// Fix delta code
-		Bit32u tdelta = delta;
-#if MT32EMU_ACCURATENOTES == 0
-		tdelta = FIXEDPOINT_UMULT(tdelta, fineShift, 12);
-#endif
-		tdelta = FIXEDPOINT_UMULT(tdelta, pitchEnvVal, 12);
-		tdelta = FIXEDPOINT_UMULT(tdelta, lfoShift, 12);
-		tdelta = FIXEDPOINT_UMULT(tdelta, bendShift, 12);
-		delta = (int)tdelta;
-
-		// Get waveform - either PCM or synthesized sawtooth or square
-		if (ampEnvVal > 0) {
-			if (patchCache->PCMPartial) {
-				// Render PCM sample
-				int ra, rb, dist;
-				Bit32u taddr;
-				Bit32u pcmAddr = pcmWave->addr;
-				if (delta < 0x10000) {
-					// Linear sound interpolation
-					taddr = pcmAddr + partialOff.pcmplace;
-					ra = synth->pcmROMData[taddr];
-					taddr++;
-					if (taddr == pcmAddr + pcmWave->len) {
-						// Past end of PCM
-						if (pcmWave->loop) {
-							rb = synth->pcmROMData[pcmAddr];
-						} else {
-							rb = 0;
-						}
-					} else {
-						rb = synth->pcmROMData[taddr];
-					}
-					dist = rb - ra;
-					sample = (ra + ((dist * (Bit32s)(partialOff.pcmoffset >> 8)) >> 8));
-				} else {
-					// Sound decimation
-					// The right way to do it is to use a lowpass filter on the waveform before selecting
-					// a point.  This is too slow.  The following approximates this as fast as possible
-					int idelta = delta >> 16;
-					taddr = pcmAddr + partialOff.pcmplace;
-					ra = synth->pcmROMData[taddr++];
-					for (int ix = 0; ix < idelta - 1; ix++) {
-						if (taddr == pcmAddr + pcmWave->len) {
-							// Past end of PCM
-							if (pcmWave->loop) {
-								taddr = pcmAddr;
-							} else {
-								// Behave as if all subsequent samples were 0
-								break;
-							}
-						}
-						ra += synth->pcmROMData[taddr++];
-					}
-					sample = ra / idelta;
-				}
-			} else {
-				// Render synthesised sample
-				int toff = partialOff.pcmplace;
-				int minorplace = partialOff.pcmoffset >> 14;
-				Bit32s filterInput;
-				Bit32s filtval = getFiltEnvelope();
-
-				//synth->printDebug("Filtval: %d", filtval);
-
-				if ((patchCache->waveform & 1) == 0) {
-					// Square waveform.  Made by combining two pregenerated bandlimited
-					// sawtooth waveforms
-					Bit32u ofsA = ((toff << 2) + minorplace) % noteLookup->waveformSize[0];
-					int width = FIXEDPOINT_UMULT(noteLookup->div2, synth->tables.pwFactor[pulsewidth], 7);
-					Bit32u ofsB = (ofsA + width) % noteLookup->waveformSize[0];
-					Bit16s pa = noteLookup->waveforms[0][ofsA];
-					Bit16s pb = noteLookup->waveforms[0][ofsB];
-					filterInput = pa - pb;
-					// Non-bandlimited squarewave
-					/*
-					ofs = FIXEDPOINT_UMULT(noteLookup->div2, synth->tables.pwFactor[patchCache->pulsewidth], 8);
-					if (toff < ofs)
-						sample = 1 * WGAMP;
-					else
-						sample = -1 * WGAMP;
-					*/
-				} else {
-					// Sawtooth.  Made by combining the full cosine and half cosine according
-					// to how it looks on the MT-32.  What it really does it takes the
-					// square wave and multiplies it by a full cosine
-					int waveoff = (toff << 2) + minorplace;
-					if (toff < noteLookup->sawTable[pulsewidth])
-						filterInput = noteLookup->waveforms[1][waveoff % noteLookup->waveformSize[1]];
-					else
-						filterInput = noteLookup->waveforms[2][waveoff % noteLookup->waveformSize[2]];
-					// This is the correct way
-					// Seems slow to me (though bandlimited) -- doesn't seem to
-					// sound any better though
-					/*
-					//int pw = (patchCache->pulsewidth * pulsemod[filtval]) >> 8;
-
-					Bit32u ofs = toff % (noteLookup->div2 >> 1);
-
-					Bit32u ofs3 = toff + FIXEDPOINT_UMULT(noteLookup->div2, synth->tables.pwFactor[patchCache->pulsewidth], 9);
-					ofs3 = ofs3 % (noteLookup->div2 >> 1);
-
-					pa = noteLookup->waveforms[0][ofs];
-					pb = noteLookup->waveforms[0][ofs3];
-					sample = ((pa - pb) * noteLookup->waveforms[2][toff]) / 2;
-					*/
-				}
-
-				//Very exact filter
-				if (filtval > ((FILTERGRAN * 15) / 16))
-					filtval = ((FILTERGRAN * 15) / 16);
-				sample = (Bit32s)(floorf((synth->iirFilter)((float)filterInput, &history[0], synth->tables.filtCoeff[filtval][(int)patchCache->filtEnv.resonance])) / synth->tables.resonanceFactor[patchCache->filtEnv.resonance]);
-				if (sample < -32768) {
-					synth->printDebug("Overdriven amplitude for %d: %d:=%d < -32768", patchCache->waveform, filterInput, sample);
-					sample = -32768;
-				}
-				else if (sample > 32767) {
-					synth->printDebug("Overdriven amplitude for %d: %d:=%d > 32767", patchCache->waveform, filterInput, sample);
-					sample = 32767;
-				}
-			}
-		}
-
-		// Add calculated delta to our waveform offset
-		Bit32u absOff = ((partialOff.pcmplace << 16) | partialOff.pcmoffset);
-		absOff += delta;
-		partialOff.pcmplace = (Bit16u)((absOff & 0xFFFF0000) >> 16);
-		partialOff.pcmoffset = (Bit16u)(absOff & 0xFFFF);
-
-		// Put volume envelope over generated sample
-		sample = FIXEDPOINT_SMULT(sample, ampEnvVal, 9);
-		sample = FIXEDPOINT_SMULT(sample, volume, 7);
-		envs[EnvelopeType_amp].envpos++;
-		envs[EnvelopeType_pitch].envpos++;
-		envs[EnvelopeType_filt].envpos++;
-
-		*partialBuf++ = (Bit16s)sample;
-	}
-	// We may have deactivated and broken out of the loop before the end of the buffer,
-	// if so then fill the remainder with 0s.
-	if (++length > 0)
-		memset(partialBuf, 0, length * 2);
-	return &myBuffer[0];
-}
-
-void Partial::setBend(float factor) {
-	if (!patchCache->useBender || factor == 0.0f) {
-		bendShift = 4096;
-		return;
-	}
-	// NOTE:KG: We can't do this smoothly with lookup tables, unless we use several MB.
-	// FIXME:KG: Bend should be influenced by pitch key-follow too, according to docs.
-	float bendSemitones = factor * patchCache->benderRange; // -24 .. 24
-	float mult = powf(2.0f, bendSemitones / 12.0f);
-	synth->printDebug("setBend(): factor=%f, benderRange=%f, semitones=%f, mult=%f\n", (double)factor, (double)patchCache->benderRange, (double)bendSemitones, (double)mult);
-	bendShift = (int)(mult * 4096.0f);
-}
-
-Bit16s *Partial::mixBuffers(Bit16s * buf1, Bit16s *buf2, int len) {
-	if (buf1 == NULL)
-		return buf2;
-	if (buf2 == NULL)
-		return buf1;
-
-	Bit16s *outBuf = buf1;
-#if MT32EMU_USE_MMX >= 1
-	// KG: This seems to be fine
-	int donelen = i386_mixBuffers(buf1, buf2, len);
-	len -= donelen;
-	buf1 += donelen;
-	buf2 += donelen;
-#endif
-	while (len--) {
-		*buf1 = *buf1 + *buf2;
-		buf1++, buf2++;
-	}
-	return outBuf;
-}
-
-Bit16s *Partial::mixBuffersRingMix(Bit16s * buf1, Bit16s *buf2, int len) {
-	if (buf1 == NULL)
-		return NULL;
-	if (buf2 == NULL) {
-		Bit16s *outBuf = buf1;
-		while (len--) {
-			if (*buf1 < -8192)
-				*buf1 = -8192;
-			else if (*buf1 > 8192)
-				*buf1 = 8192;
-			buf1++;
-		}
-		return outBuf;
-	}
-
-	Bit16s *outBuf = buf1;
-#if MT32EMU_USE_MMX >= 1
-	// KG: This seems to be fine
-	int donelen = i386_mixBuffersRingMix(buf1, buf2, len);
-	len -= donelen;
-	buf1 += donelen;
-	buf2 += donelen;
-#endif
-	while (len--) {
-		float a, b;
-		a = ((float)*buf1) / 8192.0f;
-		b = ((float)*buf2) / 8192.0f;
-		a = (a * b) + a;
-		if (a > 1.0f)
-			a = 1.0f;
-		if (a < -1.0f)
-			a = -1.0f;
-		*buf1 = (Bit16s)(a * 8192.0f);
-		buf1++;
-		buf2++;
-		//buf1[i] = (Bit16s)(((Bit32s)buf1[i] * (Bit32s)buf2[i]) >> 10) + buf1[i];
-	}
-	return outBuf;
-}
-
-Bit16s *Partial::mixBuffersRing(Bit16s * buf1, Bit16s *buf2, int len) {
-	if (buf1 == NULL) {
-		return NULL;
-	}
-	if (buf2 == NULL) {
-		return NULL;
-	}
-
-	Bit16s *outBuf = buf1;
-#if MT32EMU_USE_MMX >= 1
-	// FIXME:KG: Not really checked as working
-	int donelen = i386_mixBuffersRing(buf1, buf2, len);
-	len -= donelen;
-	buf1 += donelen;
-	buf2 += donelen;
-#endif
-	while (len--) {
-		float a, b;
-		a = ((float)*buf1) / 8192.0f;
-		b = ((float)*buf2) / 8192.0f;
-		a *= b;
-		if (a > 1.0f)
-			a = 1.0f;
-		if (a < -1.0f)
-			a = -1.0f;
-		*buf1 = (Bit16s)(a * 8192.0f);
-		buf1++;
-		buf2++;
-	}
-	return outBuf;
-}
-
-void Partial::mixBuffersStereo(Bit16s *buf1, Bit16s *buf2, Bit16s *outBuf, int len) {
-	if (buf2 == NULL) {
-		while (len--) {
-			*outBuf++ = *buf1++;
-			*outBuf++ = 0;
-		}
-	} else if (buf1 == NULL) {
-		while (len--) {
-			*outBuf++ = 0;
-			*outBuf++ = *buf2++;
-		}
-	} else {
-		while (len--) {
-			*outBuf++ = *buf1++;
-			*outBuf++ = *buf2++;
-		}
-	}
-}
-
-bool Partial::produceOutput(Bit16s *partialBuf, long length) {
-	if (!isActive() || alreadyOutputed)
-		return false;
-	if (poly == NULL) {
-		synth->printDebug("*** ERROR: poly is NULL at Partial::produceOutput()!");
-		return false;
-	}
-
-	Bit16s *pairBuf = NULL;
-	// Check for dependant partial
-	if (pair != NULL) {
-		if (!pair->alreadyOutputed) {
-			// Note: pair may have become NULL after this
-			pairBuf = pair->generateSamples(length);
-		}
-	} else if (useNoisePair) {
-		// Generate noise for pairless ring mix
-		pairBuf = synth->tables.noiseBuf;
-	}
-
-	Bit16s *myBuf = generateSamples(length);
-
-	if (myBuf == NULL && pairBuf == NULL)
-		return false;
-
-	Bit16s *p1buf, *p2buf;
-
-	if (structurePosition == 0 || pairBuf == NULL) {
-		p1buf = myBuf;
-		p2buf = pairBuf;
-	} else {
-		p2buf = myBuf;
-		p1buf = pairBuf;
-	}
-
-	//synth->printDebug("mixType: %d", mixType);
-
-	Bit16s *mixedBuf;
-	switch (mixType) {
-	case 0:
-		// Standard sound mix
-		mixedBuf = mixBuffers(p1buf, p2buf, length);
-		break;
-
-	case 1:
-		// Ring modulation with sound mix
-		mixedBuf = mixBuffersRingMix(p1buf, p2buf, length);
-		break;
-
-	case 2:
-		// Ring modulation alone
-		mixedBuf = mixBuffersRing(p1buf, p2buf, length);
-		break;
-
-	case 3:
-		// Stereo mixing.  One partial to one speaker channel, one to another.
-		// FIXME:KG: Surely we should be multiplying by the left/right volumes here?
-		mixBuffersStereo(p1buf, p2buf, partialBuf, length);
-		return true;
-
-	default:
-		mixedBuf = mixBuffers(p1buf, p2buf, length);
-		break;
-	}
-
-	if (mixedBuf == NULL)
-		return false;
-
-	Bit16s leftvol, rightvol;
-	leftvol = patchCache->pansetptr->leftvol;
-	rightvol = patchCache->pansetptr->rightvol;
-
-#if MT32EMU_USE_MMX >= 2
-	// FIXME:KG: This appears to introduce crackle
-	int donelen = i386_partialProductOutput(length, leftvol, rightvol, partialBuf, mixedBuf);
-	length -= donelen;
-	mixedBuf += donelen;
-	partialBuf += donelen * 2;
-#endif
-	while (length--) {
-		*partialBuf++ = (Bit16s)(((Bit32s)*mixedBuf * (Bit32s)leftvol) >> 15);
-		*partialBuf++ = (Bit16s)(((Bit32s)*mixedBuf * (Bit32s)rightvol) >> 15);
-		mixedBuf++;
-	}
-	return true;
-}
-
-Bit32s Partial::getFiltEnvelope() {
-	int reshigh;
-
-	int cutoff, depth;
-
-	EnvelopeStatus *tStat  = &envs[EnvelopeType_filt];
-
-	if (tStat->decaying) {
-		reshigh = tStat->envbase;
-		reshigh = (reshigh + ((tStat->envdist * tStat->envpos) / tStat->envsize));
-		if (tStat->envpos >= tStat->envsize)
-			reshigh = 0;
-	} else {
-		if (tStat->envstat==4) {
-			reshigh = patchCache->filtsustain;
-			if (!poly->sustain) {
-				startDecay(EnvelopeType_filt, reshigh);
-			}
-		} else {
-			if ((tStat->envstat==-1) || (tStat->envpos >= tStat->envsize)) {
-				if (tStat->envstat==-1)
-					tStat->envbase = 0;
-				else
-					tStat->envbase = patchCache->filtEnv.envlevel[tStat->envstat];
-				tStat->envstat++;
-				tStat->envpos = 0;
-				if (tStat->envstat == 3) {
-					tStat->envsize = synth->tables.envTime[(int)patchCache->filtEnv.envtime[tStat->envstat]];
-				} else {
-					Bit32u envTime = (int)patchCache->filtEnv.envtime[tStat->envstat];
-					if (tStat->envstat > 1) {
-						int envDiff = abs(patchCache->filtEnv.envlevel[tStat->envstat] - patchCache->filtEnv.envlevel[tStat->envstat - 1]);
-						if (envTime > synth->tables.envDeltaMaxTime[envDiff]) {
-							envTime = synth->tables.envDeltaMaxTime[envDiff];
-						}
-					}
-
-					tStat->envsize = (synth->tables.envTime[envTime] * keyLookup->envTimeMult[(int)patchCache->filtEnv.envtkf]) >> 8;
-				}
-
-				tStat->envsize++;
-				tStat->envdist = patchCache->filtEnv.envlevel[tStat->envstat] - tStat->envbase;
-			}
-
-			reshigh = tStat->envbase;
-			reshigh = (reshigh + ((tStat->envdist * tStat->envpos) / tStat->envsize));
-
-		}
-		tStat->prevlevel = reshigh;
-	}
-
-	cutoff = patchCache->filtEnv.cutoff;
-
-	//if (patchCache->waveform==1) reshigh = (reshigh * 3) >> 2;
-
-	depth = patchCache->filtEnv.envdepth;
-
-	//int sensedep = (depth * 127-patchCache->filtEnv.envsense) >> 7;
-	depth = FIXEDPOINT_UMULT(depth, synth->tables.tvfVelfollowMult[poly->vel][(int)patchCache->filtEnv.envsense], 8);
-
-	int bias = patchCache->tvfbias;
-	int dist;
-
-	if (bias != 0) {
-		//FIXME:KG: Is this really based on pitch (as now), or key pressed?
-		//synth->printDebug("Cutoff before %d", cutoff);
-		if (patchCache->tvfdir == 0) {
-			if (noteVal < bias) {
-				dist = bias - noteVal;
-				cutoff = FIXEDPOINT_UMULT(cutoff, synth->tables.tvfBiasMult[patchCache->tvfblevel][dist], 8);
-			}
-		} else {
-			// > Bias
-			if (noteVal > bias) {
-				dist = noteVal - bias;
-				cutoff = FIXEDPOINT_UMULT(cutoff, synth->tables.tvfBiasMult[patchCache->tvfblevel][dist], 8);
-			}
-
-		}
-		//synth->printDebug("Cutoff after %d", cutoff);
-	}
-
-	depth = (depth * keyLookup->envDepthMult[patchCache->filtEnv.envdkf]) >> 8;
-	reshigh = (reshigh * depth) >> 7;
-
-	Bit32s tmp;
-
-	cutoff *= filtVal;
-	cutoff /= realVal; //FIXME:KG: With filter keyfollow 0, this makes no sense. What's correct?
-
-	reshigh *= filtVal;
-	reshigh /= realVal; //FIXME:KG: As above for cutoff
-
-	if (patchCache->waveform == 1) {
-		reshigh = (reshigh * 65) / 100;
-	}
-
-	if (cutoff > 100)
-		cutoff = 100;
-	else if (cutoff < 0)
-		cutoff = 0;
-	if (reshigh > 100)
-		reshigh = 100;
-	else if (reshigh < 0)
-		reshigh = 0;
-	tmp = noteLookup->nfiltTable[cutoff][reshigh];
-	//tmp *= keyfollow;
-	//tmp /= realfollow;
-
-	//synth->printDebug("Cutoff %d, tmp %d, freq %d", cutoff, tmp, tmp * 256);
-	return tmp;
-}
-
-bool Partial::shouldReverb() {
-	if (!isActive())
-		return false;
-	return patchCache->reverb;
-}
-
-Bit32u Partial::getAmpEnvelope() {
-	Bit32s tc;
-
-	EnvelopeStatus *tStat = &envs[EnvelopeType_amp];
-
-	if (!play)
-		return 0;
-
-	if (tStat->decaying) {
-		tc = tStat->envbase;
-		tc += (tStat->envdist * tStat->envpos) / tStat->envsize;
-		if (tc < 0)
-			tc = 0;
-		if ((tStat->envpos >= tStat->envsize) || (tc == 0)) {
-			play = false;
-			// Don't have to worry about prevlevel storage or anything, this partial's about to die
-			return 0;
-		}
-	} else {
-		if ((tStat->envstat == -1) || (tStat->envpos >= tStat->envsize)) {
-			if (tStat->envstat == -1)
-				tStat->envbase = 0;
-			else
-				tStat->envbase = patchCache->ampEnv.envlevel[tStat->envstat];
-			tStat->envstat++;
-			tStat->envpos = 0;
-			if (tStat->envstat == 4) {
-				//synth->printDebug("Envstat %d, size %d", tStat->envstat, tStat->envsize);
-				tc = patchCache->ampEnv.envlevel[3];
-				if (!poly->sustain)
-					startDecay(EnvelopeType_amp, tc);
-				else
-					tStat->sustaining = true;
-				goto PastCalc;
-			}
-			Bit8u targetLevel = patchCache->ampEnv.envlevel[tStat->envstat];
-			tStat->envdist = targetLevel - tStat->envbase;
-			Bit32u envTime = patchCache->ampEnv.envtime[tStat->envstat];
-			if (targetLevel == 0) {
-				tStat->envsize = synth->tables.envDecayTime[envTime];
-			} else {
-				int envLevelDelta = abs(tStat->envdist);
-				if (envTime > synth->tables.envDeltaMaxTime[envLevelDelta]) {
-					envTime = synth->tables.envDeltaMaxTime[envLevelDelta];
-				}
-				tStat->envsize = synth->tables.envTime[envTime];
-			}
-
-			// Time keyfollow is used by all sections of the envelope (confirmed on CM-32L)
-			tStat->envsize = FIXEDPOINT_UMULT(tStat->envsize, keyLookup->envTimeMult[(int)patchCache->ampEnv.envtkf], 8);
-
-			switch (tStat->envstat) {
-			case 0:
-				//Spot for velocity time follow
-				//Only used for first attack
-				tStat->envsize = FIXEDPOINT_UMULT(tStat->envsize, synth->tables.envTimeVelfollowMult[(int)patchCache->ampEnv.envvkf][poly->vel], 8);
-				//synth->printDebug("Envstat %d, size %d", tStat->envstat, tStat->envsize);
-				break;
-			case 1:
-			case 2:
-			case 3:
-				//synth->printDebug("Envstat %d, size %d", tStat->envstat, tStat->envsize);
-				break;
-			default:
-				synth->printDebug("Invalid TVA envelope number %d hit!", tStat->envstat);
-				break;
-			}
-
-			tStat->envsize++;
-
-			if (tStat->envdist != 0) {
-				tStat->counter = abs(tStat->envsize / tStat->envdist);
-				//synth->printDebug("Pos %d, envsize %d envdist %d", tStat->envstat, tStat->envsize, tStat->envdist);
-			} else {
-				tStat->counter = 0;
-				//synth->printDebug("Pos %d, envsize %d envdist %d", tStat->envstat, tStat->envsize, tStat->envdist);
-			}
-		}
-		tc = tStat->envbase;
-		tc = (tc + ((tStat->envdist * tStat->envpos) / tStat->envsize));
-		tStat->count = tStat->counter;
-PastCalc:
-		tc = (tc * (Bit32s)patchCache->ampEnv.level) / 100;
-	}
-
-	// Prevlevel storage is bottle neck
-	tStat->prevlevel = tc;
-
-	//Bias level crap stuff now
-
-	for (int i = 0; i < 2; i++) {
-		if (patchCache->ampblevel[i]!=0) {
-			int bias = patchCache->ampbias[i];
-			if (patchCache->ampdir[i]==0) {
-				// < Bias
-				if (noteVal < bias) {
-					int dist = bias - noteVal;
-					tc = FIXEDPOINT_UMULT(tc, synth->tables.tvaBiasMult[patchCache->ampblevel[i]][dist], 8);
-				}
-			} else {
-				// > Bias
-				if (noteVal > bias) {
-					int dist = noteVal - bias;
-					tc = FIXEDPOINT_UMULT(tc, synth->tables.tvaBiasMult[patchCache->ampblevel[i]][dist], 8);
-				}
-			}
-		}
-	}
-	if (tc < 0) {
-		synth->printDebug("*** ERROR: tc < 0 (%d) at getAmpEnvelope()", tc);
-		tc = 0;
-	}
-	return (Bit32u)tc;
-}
-
-Bit32s Partial::getPitchEnvelope() {
-	EnvelopeStatus *tStat = &envs[EnvelopeType_pitch];
-
-	Bit32s tc;
-	pitchSustain = false;
-	if (tStat->decaying) {
-		if (tStat->envpos >= tStat->envsize)
-			tc = patchCache->pitchEnv.level[4];
-		else {
-			tc = tStat->envbase;
-			tc = (tc + ((tStat->envdist * tStat->envpos) / tStat->envsize));
-		}
-	} else {
-		if (tStat->envstat==3) {
-			tc = patchCache->pitchsustain;
-			if (poly->sustain)
-				pitchSustain = true;
-			else
-				startDecay(EnvelopeType_pitch, tc);
-		} else {
-			if ((tStat->envstat==-1) || (tStat->envpos >= tStat->envsize)) {
-				tStat->envstat++;
-
-				tStat->envbase = patchCache->pitchEnv.level[tStat->envstat];
-
-				Bit32u envTime = patchCache->pitchEnv.time[tStat->envstat];
-				int envDiff = abs(patchCache->pitchEnv.level[tStat->envstat] - patchCache->pitchEnv.level[tStat->envstat + 1]);
-				if (envTime > synth->tables.envDeltaMaxTime[envDiff]) {
-					envTime = synth->tables.envDeltaMaxTime[envDiff];
-				}
-
-				tStat->envsize = (synth->tables.envTime[envTime] * keyLookup->envTimeMult[(int)patchCache->pitchEnv.timekeyfollow]) >> 8;
-
-				tStat->envpos = 0;
-				tStat->envsize++;
-				tStat->envdist = patchCache->pitchEnv.level[tStat->envstat + 1] - tStat->envbase;
-			}
-			tc = tStat->envbase;
-			tc = (tc + ((tStat->envdist * tStat->envpos) / tStat->envsize));
-		}
-		tStat->prevlevel = tc;
-	}
-	return tc;
-}
-
-void Partial::startDecayAll() {
-	startDecay(EnvelopeType_amp, envs[EnvelopeType_amp].prevlevel);
-	startDecay(EnvelopeType_filt, envs[EnvelopeType_filt].prevlevel);
-	startDecay(EnvelopeType_pitch, envs[EnvelopeType_pitch].prevlevel);
-	pitchSustain = false;
-}
-
-void Partial::startDecay(EnvelopeType envnum, Bit32s startval) {
-	EnvelopeStatus *tStat  = &envs[envnum];
-
-	tStat->sustaining = false;
-	tStat->decaying = true;
-	tStat->envpos = 0;
-	tStat->envbase = startval;
-
-	switch (envnum) {
-	case EnvelopeType_amp:
-		tStat->envsize = FIXEDPOINT_UMULT(synth->tables.envDecayTime[(int)patchCache->ampEnv.envtime[4]], keyLookup->envTimeMult[(int)patchCache->ampEnv.envtkf], 8);
-		tStat->envdist = -startval;
-		break;
-	case EnvelopeType_filt:
-		tStat->envsize = FIXEDPOINT_UMULT(synth->tables.envDecayTime[(int)patchCache->filtEnv.envtime[4]], keyLookup->envTimeMult[(int)patchCache->filtEnv.envtkf], 8);
-		tStat->envdist = -startval;
-		break;
-	case EnvelopeType_pitch:
-		tStat->envsize = FIXEDPOINT_UMULT(synth->tables.envDecayTime[(int)patchCache->pitchEnv.time[3]], keyLookup->envTimeMult[(int)patchCache->pitchEnv.timekeyfollow], 8);
-		tStat->envdist = patchCache->pitchEnv.level[4] - startval;
-		break;
-	default:
-		break;
-	}
-	tStat->envsize++;
-}
diff --git a/sound/softsynth/mt32/partial.h b/sound/softsynth/mt32/partial.h
deleted file mode 100644
index 93d8bcd985..0000000000
--- a/sound/softsynth/mt32/partial.h
+++ /dev/null
@@ -1,148 +0,0 @@
-/* Copyright (c) 2003-2005 Various contributors
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to
- * deal in the Software without restriction, including without limitation the
- * rights to use, copy, modify, merge, publish, distribute, sublicense, and/or
- * sell copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
- * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
- * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS
- * IN THE SOFTWARE.
- */
-
-#ifndef MT32EMU_PARTIAL_H
-#define MT32EMU_PARTIAL_H
-
-namespace MT32Emu {
-
-class Synth;
-struct NoteLookup;
-
-enum EnvelopeType {
-	EnvelopeType_amp = 0,
-	EnvelopeType_filt = 1,
-	EnvelopeType_pitch = 2
-};
-
-struct EnvelopeStatus {
-	Bit32s envpos;
-	Bit32s envstat;
-	Bit32s envbase;
-	Bit32s envdist;
-	Bit32s envsize;
-
-	bool sustaining;
-	bool decaying;
-	Bit32s prevlevel;
-
-	Bit32s counter;
-	Bit32s count;
-};
-
-// Class definition of MT-32 partials.  32 in all.
-class Partial {
-private:
-	Synth *synth;
-
-	int ownerPart; // -1 if unassigned
-	int mixType;
-	int structurePosition; // 0 or 1 of a structure pair
-	bool useNoisePair;
-
-	Bit16s myBuffer[MAX_SAMPLE_OUTPUT];
-
-	// Keyfollowed note value
-#if MT32EMU_ACCURATENOTES == 1
-	NoteLookup noteLookupStorage;
-	float noteVal;
-#else
-	int noteVal;
-	int fineShift;
-#endif
-	const NoteLookup *noteLookup; // LUTs for this noteVal
-	const KeyLookup *keyLookup; // LUTs for the clamped (12..108) key
-
-	// Keyfollowed filter values
-	int realVal;
-	int filtVal;
-
-	// Only used for PCM partials
-	int pcmNum;
-	PCMWaveEntry *pcmWave;
-
-	int pulsewidth;
-
-	Bit32u lfoPos;
-	soundaddr partialOff;
-
-	Bit32u ampEnvVal;
-	Bit32u pitchEnvVal;
-
-	float history[32];
-
-	bool pitchSustain;
-
-	int loopPos;
-
-	dpoly *poly;
-
-	int bendShift;
-
-	Bit16s *mixBuffers(Bit16s *buf1, Bit16s *buf2, int len);
-	Bit16s *mixBuffersRingMix(Bit16s *buf1, Bit16s *buf2, int len);
-	Bit16s *mixBuffersRing(Bit16s *buf1, Bit16s *buf2, int len);
-	void mixBuffersStereo(Bit16s *buf1, Bit16s *buf2, Bit16s *outBuf, int len);
-
-	Bit32s getFiltEnvelope();
-	Bit32u getAmpEnvelope();
-	Bit32s getPitchEnvelope();
-
-	void initKeyFollow(int freqNum);
-
-public:
-	const PatchCache *patchCache;
-	EnvelopeStatus envs[3];
-	bool play;
-
-	PatchCache cachebackup;
-
-	Partial *pair;
-	bool alreadyOutputed;
-	Bit32u age;
-
-	Partial(Synth *synth);
-	~Partial();
-
-	int getOwnerPart() const;
-	int getKey() const;
-	const dpoly *getDpoly() const;
-	bool isActive();
-	void activate(int part);
-	void deactivate(void);
-	void startPartial(dpoly *usePoly, const PatchCache *useCache, Partial *pairPartial);
-	void startDecay(EnvelopeType envnum, Bit32s startval);
-	void startDecayAll();
-	void setBend(float factor);
-	bool shouldReverb();
-
-	// Returns true only if data written to buffer
-	// This function (unlike the one below it) returns processed stereo samples
-	// made from combining this single partial with its pair, if it has one.
-	bool produceOutput(Bit16s * partialBuf, long length);
-
-	// This function produces mono sample output using the partial's private internal buffer
-	Bit16s *generateSamples(long length);
-};
-
-}
-
-#endif
diff --git a/sound/softsynth/mt32/partialManager.cpp b/sound/softsynth/mt32/partialManager.cpp
deleted file mode 100644
index 3d3b6302db..0000000000
--- a/sound/softsynth/mt32/partialManager.cpp
+++ /dev/null
@@ -1,272 +0,0 @@
-/* Copyright (c) 2003-2005 Various contributors
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to
- * deal in the Software without restriction, including without limitation the
- * rights to use, copy, modify, merge, publish, distribute, sublicense, and/or
- * sell copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
- * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
- * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS
- * IN THE SOFTWARE.
- */
-
-#include <string.h>
-
-#include "mt32emu.h"
-
-using namespace MT32Emu;
-
-PartialManager::PartialManager(Synth *useSynth) {
-	this->synth = useSynth;
-	for (int i = 0; i < MT32EMU_MAX_PARTIALS; i++)
-		partialTable[i] = new Partial(synth);
-}
-
-PartialManager::~PartialManager(void) {
-	for (int i = 0; i < MT32EMU_MAX_PARTIALS; i++)
-		delete partialTable[i];
-}
-
-void PartialManager::getPerPartPartialUsage(int usage[9]) {
-	memset(usage, 0, 9 * sizeof (int));
-	for (int i = 0; i < MT32EMU_MAX_PARTIALS; i++) {
-		if (partialTable[i]->isActive())
-			usage[partialTable[i]->getOwnerPart()]++;
-	}
-}
-
-void PartialManager::clearAlreadyOutputed() {
-	for (int i = 0; i < MT32EMU_MAX_PARTIALS; i++)
-		partialTable[i]->alreadyOutputed = false;
-}
-
-void PartialManager::ageAll() {
-	for (int i = 0; i < MT32EMU_MAX_PARTIALS; i++)
-		partialTable[i]->age++;
-}
-
-bool PartialManager::shouldReverb(int i) {
-	return partialTable[i]->shouldReverb();
-}
-
-bool PartialManager::produceOutput(int i, Bit16s *buffer, Bit32u bufferLength) {
-	return partialTable[i]->produceOutput(buffer, bufferLength);
-}
-
-void PartialManager::deactivateAll() {
-	for (int i = 0; i < MT32EMU_MAX_PARTIALS; i++) {
-		partialTable[i]->deactivate();
-	}
-}
-
-unsigned int PartialManager::setReserve(Bit8u *rset) {
-	unsigned int pr = 0;
-	for (int x = 0; x < 9; x++) {
-		for (int y = 0; y < rset[x]; y++) {
-			partialReserveTable[pr] = x;
-			pr++;
-		}
-	}
-	return pr;
-}
-
-Partial *PartialManager::allocPartial(int partNum) {
-	Partial *outPartial = NULL;
-
-	// Use the first inactive partial reserved for the specified part (if there are any)
-	// Otherwise, use the last inactive partial, if any
-	for (int i = 0; i < MT32EMU_MAX_PARTIALS; i++) {
-		if (!partialTable[i]->isActive()) {
-			outPartial = partialTable[i];
-			if (partialReserveTable[i] == partNum)
-				break;
-		}
-	}
-	if (outPartial != NULL) {
-		outPartial->activate(partNum);
-		outPartial->age = 0;
-	}
-	return outPartial;
-}
-
-unsigned int PartialManager::getFreePartialCount(void) {
-	int count = 0;
-	memset(partialPart, 0, sizeof(partialPart));
-	for (int i = 0; i < MT32EMU_MAX_PARTIALS; i++) {
-		if (!partialTable[i]->isActive())
-			count++;
-		else
-			partialPart[partialTable[i]->getOwnerPart()]++;
-	}
-	return count;
-}
-
-/*
-bool PartialManager::freePartials(unsigned int needed, int partNum) {
-	int i;
-	int myPartPrior = (int)mt32ram.system.reserveSettings[partNum];
-	if (myPartPrior<partialPart[partNum]) {
-		//This can have more parts, must kill off those with less priority
-		int most, mostPart;
-		while (needed > 0) {
-			int selectPart = -1;
-			//Find the worst offender with more partials than allocated and kill them
-			most = -1;
-			mostPart = -1;
-			int diff;
-
-			for (i=0;i<9;i++) {
-				diff = partialPart[i] - (int)mt32ram.system.reserveSettings[i];
-
-				if (diff>0) {
-					if (diff>most) {
-						most = diff;
-						mostPart = i;
-					}
-				}
-			}
-			selectPart = mostPart;
-			if (selectPart == -1) {
-				// All parts are within the allocated limits, you suck
-				// Look for first partial not of this part that's decaying perhaps?
-				return false;
-			}
-			bool found;
-			int oldest;
-			int oldnum;
-			while (partialPart[selectPart] > (int)mt32ram.system.reserveSettings[selectPart]) {
-				oldest = -1;
-				oldnum = -1;
-				found = false;
-				for (i = 0; i < MT32EMU_MAX_PARTIALS; i++) {
-					if (partialTable[i]->isActive) {
-						if (partialTable[i]->ownerPart == selectPart) {
-							found = true;
-							if (partialTable[i]->age > oldest) {
-								oldest = partialTable[i]->age;
-								oldnum = i;
-							}
-						}
-					}
-				}
-				if (!found) break;
-				partialTable[oldnum]->deactivate();
-				--partialPart[selectPart];
-				--needed;
-			}
-
-		}
-		return true;
-
-	} else {
-		//This part has reached its max, must kill off its own
-		bool found;
-		int oldest;
-		int oldnum;
-		while (needed > 0) {
-			oldest = -1;
-			oldnum = -1;
-			found = false;
-			for (i = 0; i < MT32EMU_MAX_PARTIALS; i++) {
-				if (partialTable[i]->isActive) {
-					if (partialTable[i]->ownerPart == partNum) {
-						found = true;
-						if (partialTable[i]->age > oldest) {
-							oldest = partialTable[i]->age;
-							oldnum = i;
-						}
-					}
-				}
-			}
-			if (!found) break;
-			partialTable[oldnum]->deactivate();
-			--needed;
-		}
-		// Couldn't free enough partials, sorry
-		if (needed>0) return false;
-		return true;
-	}
-
-}
-*/
-bool PartialManager::freePartials(unsigned int needed, int partNum) {
-	if (needed == 0) {
-		return true;
-	}
-	// Reclaim partials reserved for this part
-	// Kill those that are already decaying first
-	/*
-	for (int i = 0; i < MT32EMU_MAX_PARTIALS; i++) {
-		if (partialReserveTable[i] == partNum) {
-			if (partialTable[i]->ownerPart != partNum) {
-				if (partialTable[i]->partCache->envs[AMPENV].decaying) {
-					partialTable[i]->isActive = false;
-					--needed;
-					if (needed == 0)
-						return true;
-				}
-			}
-		}
-	}*/
-	// Then kill those with the lowest part priority -- oldest at the moment
-	while (needed > 0) {
-		Bit32u prior = 0;
-		int priornum = -1;
-
-		for (int i = 0; i < MT32EMU_MAX_PARTIALS; i++) {
-			if (partialReserveTable[i] == partNum && partialTable[i]->isActive() && partialTable[i]->getOwnerPart() != partNum) {
-				/*
-				if (mt32ram.system.reserveSettings[partialTable[i]->ownerPart] < prior) {
-					prior = mt32ram.system.reserveSettings[partialTable[i]->ownerPart];
-					priornum = i;
-				}*/
-				if (partialTable[i]->age >= prior) {
-					prior = partialTable[i]->age;
-					priornum = i;
-				}
-			}
-		}
-		if (priornum != -1) {
-			partialTable[priornum]->deactivate();
-			--needed;
-		} else {
-			break;
-		}
-	}
-
-	// Kill off the oldest partials within this part
-	while (needed > 0) {
-		Bit32u oldest = 0;
-		int oldlist = -1;
-		for (int i = 0; i < MT32EMU_MAX_PARTIALS; i++) {
-			if (partialTable[i]->getOwnerPart() == partNum && partialTable[i]->isActive()) {
-				if (partialTable[i]->age >= oldest) {
-					oldest = partialTable[i]->age;
-					oldlist = i;
-				}
-			}
-		}
-		if (oldlist != -1) {
-			partialTable[oldlist]->deactivate();
-			--needed;
-		} else {
-			break;
-		}
-	}
-	return needed == 0;
-}
-
-const Partial *PartialManager::getPartial(unsigned int partialNum) const {
-	if (partialNum > MT32EMU_MAX_PARTIALS - 1)
-		return NULL;
-	return partialTable[partialNum];
-}
diff --git a/sound/softsynth/mt32/partialManager.h b/sound/softsynth/mt32/partialManager.h
deleted file mode 100644
index b10f93ff02..0000000000
--- a/sound/softsynth/mt32/partialManager.h
+++ /dev/null
@@ -1,56 +0,0 @@
-/* Copyright (c) 2003-2005 Various contributors
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to
- * deal in the Software without restriction, including without limitation the
- * rights to use, copy, modify, merge, publish, distribute, sublicense, and/or
- * sell copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
- * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
- * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS
- * IN THE SOFTWARE.
- */
-
-#ifndef MT32EMU_PARTIALMANAGER_H
-#define MT32EMU_PARTIALMANAGER_H
-
-namespace MT32Emu {
-
-class Synth;
-
-class PartialManager {
-private:
-	Synth *synth; // Only used for sending debug output
-
-	Partial *partialTable[MT32EMU_MAX_PARTIALS];
-	Bit32s partialReserveTable[MT32EMU_MAX_PARTIALS];
-	Bit32s partialPart[9]; // The count of partials played per part
-
-public:
-
-	PartialManager(Synth *synth);
-	~PartialManager();
-	Partial *allocPartial(int partNum);
-	unsigned int getFreePartialCount(void);
-	bool freePartials(unsigned int needed, int partNum);
-	unsigned int setReserve(Bit8u *rset);
-	void deactivateAll();
-	void ageAll();
-	bool produceOutput(int i, Bit16s *buffer, Bit32u bufferLength);
-	bool shouldReverb(int i);
-	void clearAlreadyOutputed();
-	void getPerPartPartialUsage(int usage[9]);
-	const Partial *getPartial(unsigned int partialNum) const;
-};
-
-}
-
-#endif
diff --git a/sound/softsynth/mt32/structures.h b/sound/softsynth/mt32/structures.h
deleted file mode 100644
index ef58c1d20f..0000000000
--- a/sound/softsynth/mt32/structures.h
+++ /dev/null
@@ -1,284 +0,0 @@
-/* Copyright (c) 2003-2005 Various contributors
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to
- * deal in the Software without restriction, including without limitation the
- * rights to use, copy, modify, merge, publish, distribute, sublicense, and/or
- * sell copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
- * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
- * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS
- * IN THE SOFTWARE.
- */
-
-#ifndef MT32EMU_STRUCTURES_H
-#define MT32EMU_STRUCTURES_H
-
-namespace MT32Emu {
-
-const unsigned int MAX_SAMPLE_OUTPUT = 4096;
-
-// MT32EMU_MEMADDR() converts from sysex-padded, MT32EMU_SYSEXMEMADDR converts to it
-// Roland provides documentation using the sysex-padded addresses, so we tend to use that in code and output
-#define MT32EMU_MEMADDR(x) ((((x) & 0x7f0000) >> 2) | (((x) & 0x7f00) >> 1) | ((x) & 0x7f))
-#define MT32EMU_SYSEXMEMADDR(x) ((((x) & 0x1FC000) << 2) | (((x) & 0x3F80) << 1) | ((x) & 0x7f))
-
-#ifdef _MSC_VER
-#define  MT32EMU_ALIGN_PACKED __declspec(align(1))
-typedef unsigned __int64   Bit64u;
-typedef   signed __int64   Bit64s;
-#else
-#define MT32EMU_ALIGN_PACKED __attribute__((packed))
-typedef unsigned long long Bit64u;
-typedef   signed long long Bit64s;
-#endif
-
-typedef unsigned int       Bit32u;
-typedef   signed int       Bit32s;
-typedef unsigned short int Bit16u;
-typedef   signed short int Bit16s;
-typedef unsigned char      Bit8u;
-typedef   signed char      Bit8s;
-
-// The following structures represent the MT-32's memory
-// Since sysex allows this memory to be written to in blocks of bytes,
-// we keep this packed so that we can copy data into the various
-// banks directly
-#if defined(_MSC_VER) || defined (__MINGW32__)
-#pragma pack(push, 1)
-#else
-#pragma pack(1)
-#endif
-
-struct TimbreParam {
-	struct commonParam {
-		char name[10];
-		Bit8u pstruct12;  // 1&2  0-12 (1-13)
-		Bit8u pstruct34;  // #3&4  0-12 (1-13)
-		Bit8u pmute;  // 0-15 (0000-1111)
-		Bit8u nosustain; // 0-1(Normal, No sustain)
-	} MT32EMU_ALIGN_PACKED common;
-
-	struct partialParam {
-		struct wgParam {
-			Bit8u coarse;  // 0-96 (C1,C#1-C9)
-			Bit8u fine;  // 0-100 (-50 to +50 (cents?))
-			Bit8u keyfollow;  // 0-16 (-1,-1/2,0,1,1/8,1/4,3/8,1/2,5/8,3/4,7/8,1,5/4,3/2,2.s1,s2)
-			Bit8u bender;  // 0,1 (ON/OFF)
-			Bit8u waveform; // MT-32: 0-1 (SQU/SAW); LAPC-I: WG WAVEFORM/PCM BANK 0 - 3 (SQU/1, SAW/1, SQU/2, SAW/2)
-			Bit8u pcmwave; // 0-127 (1-128)
-			Bit8u pulsewid; // 0-100
-			Bit8u pwvelo; // 0-14 (-7 - +7)
-		} MT32EMU_ALIGN_PACKED wg;
-
-		struct envParam {
-			Bit8u depth; // 0-10
-			Bit8u sensitivity; // 1-100
-			Bit8u timekeyfollow; // 0-4
-			Bit8u time[4]; // 1-100
-			Bit8u level[5]; // 1-100 (-50 - +50)
-		} MT32EMU_ALIGN_PACKED env;
-
-		struct lfoParam {
-			Bit8u rate; // 0-100
-			Bit8u depth; // 0-100
-			Bit8u modsense; // 0-100
-		} MT32EMU_ALIGN_PACKED lfo;
-
-		struct tvfParam {
-			Bit8u cutoff; // 0-100
-			Bit8u resonance; // 0-30
-			Bit8u keyfollow; // 0-16 (-1,-1/2,1/4,0,1,1/8,1/4,3/8,1/2,5/8,3/2,7/8,1,5/4,3/2,2,s1,s2)
-			Bit8u biaspoint; // 0-127 (<1A-<7C >1A-7C)
-			Bit8u biaslevel; // 0-14 (-7 - +7)
-			Bit8u envdepth; // 0-100
-			Bit8u envsense; // 0-100
-			Bit8u envdkf; // DEPTH KEY FOLL0W 0-4
-			Bit8u envtkf; // TIME KEY FOLLOW 0-4
-			Bit8u envtime[5]; // 1-100
-			Bit8u envlevel[4]; // 1-100
-		} MT32EMU_ALIGN_PACKED tvf;
-
-		struct tvaParam {
-			Bit8u level; // 0-100
-			Bit8u velosens; // 0-100
-			Bit8u biaspoint1; // 0-127 (<1A-<7C >1A-7C)
-			Bit8u biaslevel1; // 0-12 (-12 - 0)
-			Bit8u biaspoint2; // 0-127 (<1A-<7C >1A-7C)
-			Bit8u biaslevel2; // 0-12 (-12 - 0)
-			Bit8u envtkf; // TIME KEY FOLLOW 0-4
-			Bit8u envvkf; // VELOS KEY FOLL0W 0-4
-			Bit8u envtime[5]; // 1-100
-			Bit8u envlevel[4]; // 1-100
-		} MT32EMU_ALIGN_PACKED tva;
-	} MT32EMU_ALIGN_PACKED partial[4];
-} MT32EMU_ALIGN_PACKED;
-
-struct PatchParam {
-	Bit8u timbreGroup; // TIMBRE GROUP  0-3 (group A, group B, Memory, Rhythm)
-	Bit8u timbreNum; // TIMBRE NUMBER 0-63
-	Bit8u keyShift; // KEY SHIFT 0-48 (-24 - +24 semitones)
-	Bit8u fineTune; // FINE TUNE 0-100 (-50 - +50 cents)
-	Bit8u benderRange; // BENDER RANGE 0-24
-	Bit8u assignMode;  // ASSIGN MODE 0-3 (POLY1, POLY2, POLY3, POLY4)
-	Bit8u reverbSwitch;  // REVERB SWITCH 0-1 (OFF,ON)
-	Bit8u dummy; // (DUMMY)
-} MT32EMU_ALIGN_PACKED;
-
-struct MemParams {
-	// NOTE: The MT-32 documentation only specifies PatchTemp areas for parts 1-8.
-	// The LAPC-I documentation specified an additional area for rhythm at the end,
-	// where all parameters but fine tune, assign mode and output level are ignored
-	struct PatchTemp {
-		PatchParam patch;
-		Bit8u outlevel; // OUTPUT LEVEL 0-100
-		Bit8u panpot; // PANPOT 0-14 (R-L)
-		Bit8u dummyv[6];
-	} MT32EMU_ALIGN_PACKED;
-
-	PatchTemp patchSettings[9];
-
-	struct RhythmTemp {
-		Bit8u timbre; // TIMBRE  0-94 (M1-M64,R1-30,OFF)
-		Bit8u outlevel; // OUTPUT LEVEL 0-100
-		Bit8u panpot; // PANPOT 0-14 (R-L)
-		Bit8u reverbSwitch;  // REVERB SWITCH 0-1 (OFF,ON)
-	} MT32EMU_ALIGN_PACKED;
-
-	RhythmTemp rhythmSettings[85];
-
-	TimbreParam timbreSettings[8];
-
-	PatchParam patches[128];
-
-	// NOTE: There are only 30 timbres in the "rhythm" bank for MT-32; the additional 34 are for LAPC-I and above
-	struct PaddedTimbre {
-		TimbreParam timbre;
-		Bit8u padding[10];
-	} MT32EMU_ALIGN_PACKED;
-
-	PaddedTimbre timbres[64 + 64 + 64 + 64]; // Group A, Group B, Memory, Rhythm
-
-	struct SystemArea {
-		Bit8u masterTune; // MASTER TUNE 0-127 432.1-457.6Hz
-		Bit8u reverbMode; // REVERB MODE 0-3 (room, hall, plate, tap delay)
-		Bit8u reverbTime; // REVERB TIME 0-7 (1-8)
-		Bit8u reverbLevel; // REVERB LEVEL 0-7 (1-8)
-		Bit8u reserveSettings[9]; // PARTIAL RESERVE (PART 1) 0-32
-		Bit8u chanAssign[9]; // MIDI CHANNEL (PART1) 0-16 (1-16,OFF)
-		Bit8u masterVol; // MASTER VOLUME 0-100
-	} MT32EMU_ALIGN_PACKED;
-
-	SystemArea system;
-};
-
-#if defined(_MSC_VER) || defined (__MINGW32__)
-#pragma pack(pop)
-#else
-#pragma pack()
-#endif
-
-struct PCMWaveEntry {
-	Bit32u addr;
-	Bit32u len;
-	double tune;
-	bool loop;
-};
-
-struct soundaddr {
-	Bit16u pcmplace;
-	Bit16u pcmoffset;
-};
-
-struct StereoVolume {
-	Bit16s leftvol;
-	Bit16s rightvol;
-};
-
-// This is basically a per-partial, pre-processed combination of timbre and patch/rhythm settings
-struct PatchCache {
-	bool playPartial;
-	bool PCMPartial;
-	int pcm;
-	char waveform;
-	int pulsewidth;
-	int pwsens;
-
-	float pitch;
-
-	int lfodepth;
-	int lforate;
-	Bit32u lfoperiod;
-	int modsense;
-
-	float pitchKeyfollow;
-
-	int filtkeyfollow;
-
-	int tvfbias;
-	int tvfblevel;
-	int tvfdir;
-
-	int ampbias[2];
-	int ampblevel[2];
-	int ampdir[2];
-
-	int ampdepth;
-	int amplevel;
-
-	bool useBender;
-	float benderRange; // 0.0, 1.0, .., 24.0 (semitones)
-
-	TimbreParam::partialParam::envParam pitchEnv;
-	TimbreParam::partialParam::tvaParam ampEnv;
-	TimbreParam::partialParam::tvfParam filtEnv;
-
-	Bit32s pitchsustain;
-	Bit32s filtsustain;
-
-	Bit32u structureMix;
-	int structurePosition;
-	int structurePair;
-
-	// The following fields are actually common to all partials in the timbre
-	bool dirty;
-	Bit32u partialCount;
-	bool sustain;
-	float pitchShift;
-	bool reverb;
-	const StereoVolume *pansetptr;
-};
-
-class Partial; // Forward reference for class defined in partial.h
-
-struct dpoly {
-	bool isPlaying;
-
-	unsigned int key;
-	int freqnum;
-	int vel;
-
-	bool isDecay;
-
-	const Bit32u *volumeptr;
-
-	Partial *partials[4];
-
-	bool pedalhold; // This marks keys that have been released on the keyboard, but are being held by the pedal
-	bool sustain;
-
-	bool isActive() const;
-	Bit32u getAge() const;
-};
-
-}
-
-#endif
diff --git a/sound/softsynth/mt32/synth.cpp b/sound/softsynth/mt32/synth.cpp
deleted file mode 100644
index 16460795a5..0000000000
--- a/sound/softsynth/mt32/synth.cpp
+++ /dev/null
@@ -1,1198 +0,0 @@
-/* Copyright (c) 2003-2005 Various contributors
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to
- * deal in the Software without restriction, including without limitation the
- * rights to use, copy, modify, merge, publish, distribute, sublicense, and/or
- * sell copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
- * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
- * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS
- * IN THE SOFTWARE.
- */
-
-#include <math.h>
-#include <string.h>
-#include <stdlib.h>
-
-#include "mt32emu.h"
-
-#include "common/str.h"
-
-#if defined(MACOSX) || defined(SOLARIS) || defined(__MINGW32__)
-// Older versions of Mac OS X didn't supply a powf function, so using it
-// will cause a binary incompatibility when trying to run a binary built
-// on a newer OS X release on an olderr one. And Solaris 8 doesn't provide
-// powf, floorf, fabsf etc. at all.
-// Cross-compiled MinGW32 toolchains suffer from a cross-compile bug in
-// libstdc++. math/stubs.o should be empty, but it comes with a symbol for
-// powf, resulting in a linker error because of multiple definitions.
-// Hence we re-define them here. The only potential drawback is that it
-// might be a little bit slower this way.
-#define powf(x,y)	((float)pow(x,y))
-#define floorf(x)	((float)floor(x))
-#define fabsf(x)	((float)fabs(x))
-#endif
-
-namespace MT32Emu {
-
-const int MAX_SYSEX_SIZE = 512;
-
-const ControlROMMap ControlROMMaps[5] = {
-	// ID    IDc IDbytes                     PCMmap  PCMc  tmbrA   tmbrAO, tmbrB   tmbrBO, tmbrR   trC  rhythm  rhyC  rsrv    panpot  prog
-	{0x4014, 22, "\000 ver1.04 14 July 87 ", 0x3000,  128, 0x8000, 0x0000, 0xC000, 0x4000, 0x3200,  30, 0x73A6,  85,  0x57C7, 0x57D0, 0x57E2}, // MT-32 revision 0
-	{0x4014, 22, "\000 ver1.06 31 Aug, 87 ", 0x3000,  128, 0x8000, 0x0000, 0xC000, 0x4000, 0x3200,  30, 0x7414,  85,  0x57D9, 0x57E2, 0x57F4}, // MT-32 revision 0
-	{0x4010, 22, "\000 ver1.07 10 Oct, 87 ", 0x3000,  128, 0x8000, 0x0000, 0xC000, 0x4000, 0x3200,  30, 0x73fe,  85,  0x57B1, 0x57BA, 0x57CC}, // MT-32 revision 1
-	{0x4010, 22, "\000verX.XX  30 Sep, 88 ", 0x3000,  128, 0x8000, 0x0000, 0xC000, 0x4000, 0x3200,  30, 0x741C,  85,  0x57E5, 0x57EE, 0x5800}, // MT-32 Blue Ridge mod
-	{0x2205, 22, "\000CM32/LAPC1.02 891205", 0x8100,  256, 0x8000, 0x8000, 0x8080, 0x8000, 0x8500,  64, 0x8580,  85,  0x4F93, 0x4F9C, 0x4FAE}  // CM-32L
-	// (Note that all but CM-32L ROM actually have 86 entries for rhythmTemp)
-};
-
-float iir_filter_normal(float input, float *hist1_ptr, float *coef_ptr) {
-	float *hist2_ptr;
-	float output,new_hist;
-
-	hist2_ptr = hist1_ptr + 1; // next history
-
-	// 1st number of coefficients array is overall input scale factor, or filter gain
-	output = input * (*coef_ptr++);
-
-	output = output - *hist1_ptr * (*coef_ptr++);
-	new_hist = output - *hist2_ptr * (*coef_ptr++); // poles
-
-	output = new_hist + *hist1_ptr * (*coef_ptr++);
-	output = output + *hist2_ptr * (*coef_ptr++);   // zeros
-
-	*hist2_ptr++ = *hist1_ptr;
-	*hist1_ptr++ = new_hist;
-	hist1_ptr++;
-	hist2_ptr++;
-
-	// i = 1
-	output = output - *hist1_ptr * (*coef_ptr++);
-	new_hist = output - *hist2_ptr * (*coef_ptr++); // poles
-
-	output = new_hist + *hist1_ptr * (*coef_ptr++);
-	output = output + *hist2_ptr * (*coef_ptr++);   // zeros
-
-	*hist2_ptr++ = *hist1_ptr;
-	*hist1_ptr++ = new_hist;
-
-	return(output);
-}
-
-Bit8u Synth::calcSysexChecksum(const Bit8u *data, Bit32u len, Bit8u checksum) {
-	for (unsigned int i = 0; i < len; i++) {
-		checksum = checksum + data[i];
-	}
-	checksum = checksum & 0x7f;
-	if (checksum)
-		checksum = 0x80 - checksum;
-	return checksum;
-}
-
-Synth::Synth() {
-	isOpen = false;
-	reverbModel = NULL;
-	partialManager = NULL;
-	memset(parts, 0, sizeof(parts));
-}
-
-Synth::~Synth() {
-	close(); // Make sure we're closed and everything is freed
-}
-
-int Synth::report(ReportType type, const void *data) {
-	if (myProp.report != NULL) {
-		return myProp.report(myProp.userData, type, data);
-	}
-	return 0;
-}
-
-void Synth::printDebug(const char *fmt, ...) {
-	va_list ap;
-	va_start(ap, fmt);
-	if (myProp.printDebug != NULL) {
-		myProp.printDebug(myProp.userData, fmt, ap);
-	} else {
-		vprintf(fmt, ap);
-		printf("\n");
-	}
-	va_end(ap);
-}
-
-void Synth::initReverb(Bit8u newRevMode, Bit8u newRevTime, Bit8u newRevLevel) {
-	// FIXME:KG: I don't think it's necessary to recreate the reverbModel... Just set the parameters
-	delete reverbModel;
-	reverbModel = new revmodel();
-
-	switch (newRevMode) {
-	case 0:
-		reverbModel->setroomsize(.1f);
-		reverbModel->setdamp(.75f);
-		break;
-	case 1:
-		reverbModel->setroomsize(.5f);
-		reverbModel->setdamp(.5f);
-		break;
-	case 2:
-		reverbModel->setroomsize(.5f);
-		reverbModel->setdamp(.1f);
-		break;
-	case 3:
-		reverbModel->setroomsize(1.0f);
-		reverbModel->setdamp(.75f);
-		break;
-	default:
-		reverbModel->setroomsize(.1f);
-		reverbModel->setdamp(.5f);
-		break;
-	}
-	reverbModel->setdry(1);
-	reverbModel->setwet((float)newRevLevel / 8.0f);
-	reverbModel->setwidth((float)newRevTime / 8.0f);
-}
-
-File *Synth::openFile(const char *filename, File::OpenMode mode) {
-	// It should never happen that openFile is NULL in our use case.
-	// Just to cover the case where something is horrible wrong we
-	// use an assert here.
-	assert(myProp.openFile != NULL);
-	return myProp.openFile(myProp.userData, filename, mode);
-}
-
-void Synth::closeFile(File *file) {
-	if (myProp.closeFile != NULL) {
-		myProp.closeFile(myProp.userData, file);
-	} else {
-		file->close();
-		delete file;
-	}
-}
-
-bool Synth::loadPreset(File *file) {
-	bool inSys = false;
-	Bit8u sysexBuf[MAX_SYSEX_SIZE];
-	Bit16u syslen = 0;
-	bool rc = true;
-	for (;;) {
-		Bit8u c;
-		if (!file->readBit8u(&c)) {
-			if (!file->isEOF()) {
-				rc = false;
-			}
-			break;
-		}
-		sysexBuf[syslen] = c;
-		if (inSys) {
-			syslen++;
-			if (c == 0xF7) {
-				playSysex(&sysexBuf[0], syslen);
-				inSys = false;
-				syslen = 0;
-			} else if (syslen == MAX_SYSEX_SIZE) {
-				printDebug("MAX_SYSEX_SIZE (%d) exceeded while processing preset, ignoring message", MAX_SYSEX_SIZE);
-				inSys = false;
-				syslen = 0;
-			}
-		} else if (c == 0xF0) {
-			syslen++;
-			inSys = true;
-		}
-	}
-	return rc;
-}
-
-bool Synth::loadControlROM(const char *filename) {
-	File *file = openFile(filename, File::OpenMode_read); // ROM File
-	if (file == NULL) {
-		return false;
-	}
-	bool rc = (file->read(controlROMData, CONTROL_ROM_SIZE) == CONTROL_ROM_SIZE);
-
-	closeFile(file);
-	if (!rc)
-		return rc;
-
-	// Control ROM successfully loaded, now check whether it's a known type
-	controlROMMap = NULL;
-	for (unsigned int i = 0; i < sizeof (ControlROMMaps) / sizeof (ControlROMMaps[0]); i++) {
-		if (memcmp(&controlROMData[ControlROMMaps[i].idPos], ControlROMMaps[i].idBytes, ControlROMMaps[i].idLen) == 0) {
-			controlROMMap = &ControlROMMaps[i];
-			return true;
-		}
-	}
-	return false;
-}
-
-bool Synth::loadPCMROM(const char *filename) {
-	File *file = openFile(filename, File::OpenMode_read); // ROM File
-	if (file == NULL) {
-		return false;
-	}
-	bool rc = true;
-	int i;
-	for (i = 0; i < pcmROMSize; i++) {
-		Bit8u s;
-		if (!file->readBit8u(&s)) {
-			if (!file->isEOF()) {
-				rc = false;
-			}
-			break;
-		}
-		Bit8u c;
-		if (!file->readBit8u(&c)) {
-			if (!file->isEOF()) {
-				rc = false;
-			} else {
-				printDebug("PCM ROM file has an odd number of bytes! Ignoring last");
-			}
-			break;
-		}
-
-		short e;
-		int bit;
-		int u;
-		int order[16] = {0, 9, 1 ,2, 3, 4, 5, 6, 7, 10, 11, 12, 13, 14, 15, 8};
-
-		e = 0;
-		for (u = 0; u < 15; u++) {
-			if (order[u] < 8)
-				bit = (s >> (7 - order[u])) & 0x1;
-			else
-				bit = (c >> (7  - (order[u] - 8))) & 0x1;
-			e = e | (short)(bit << (15 - u));
-		}
-
-		/*
-		//Bit16s e = (  ((s & 0x7f) << 4) | ((c & 0x40) << 6) | ((s & 0x80) << 6) | ((c & 0x3f))) << 2;
-		if (e<0)
-			e = -32767 - e;
-		int ut = abs(e);
-		int dif = 0x7fff - ut;
-		x = exp(((float)((float)0x8000-(float)dif) / (float)0x1000));
-		e = (int)((float)e * (x/3200));
-		*/
-
-		// File is companded (dB?), convert to linear PCM
-		// MINDB = -96
-		// MAXDB = -15
-		float testval;
-		testval = (float)((~e) & 0x7fff);
-		testval = -(testval / 400.00f);
-		//testval = -(testval / 341.32291666666666666666666666667);
-		float vol = powf(8, testval / 20) * 32767.0f;
-
-		if (e > 0)
-			vol = -vol;
-
-		pcmROMData[i] = (Bit16s)vol;
-	}
-	if (i != pcmROMSize) {
-		printDebug("PCM ROM file is too short (expected %d, got %d)", pcmROMSize, i);
-		rc = false;
-	}
-	closeFile(file);
-	return rc;
-}
-
-bool Synth::initPCMList(Bit16u mapAddress, Bit16u count) {
-	ControlROMPCMStruct *tps = (ControlROMPCMStruct *)&controlROMData[mapAddress];
-	for (int i = 0; i < count; i++) {
-		int rAddr = tps[i].pos * 0x800;
-		int rLenExp = (tps[i].len & 0x70) >> 4;
-		int rLen = 0x800 << rLenExp;
-		bool rLoop = (tps[i].len & 0x80) != 0;
-		//Bit8u rFlag = tps[i].len & 0x0F;
-		Bit16u rTuneOffset = (tps[i].pitchMSB << 8) | tps[i].pitchLSB;
-		// The number below is confirmed to a reasonable degree of accuracy on CM-32L
-		double STANDARDFREQ = 442.0;
-		float rTune = (float)(STANDARDFREQ * pow(2.0, (0x5000 - rTuneOffset) / 4056.0 - 9.0 / 12.0));
-		//printDebug("%f,%d,%d", (double)pTune, tps[i].pitchCoarse, tps[i].pitchFine);
-		if (rAddr + rLen > pcmROMSize) {
-			printDebug("Control ROM error: Wave map entry %d points to invalid PCM address 0x%04X, length 0x%04X", i, rAddr, rLen);
-			return false;
-		}
-		pcmWaves[i].addr = rAddr;
-		pcmWaves[i].len = rLen;
-		pcmWaves[i].loop = rLoop;
-		pcmWaves[i].tune = rTune;
-	}
-	return false;
-}
-
-bool Synth::initRhythmTimbre(int timbreNum, const Bit8u *mem, unsigned int memLen) {
-	if (memLen < sizeof(TimbreParam::commonParam)) {
-		return false;
-	}
-	TimbreParam *timbre = &mt32ram.timbres[timbreNum].timbre;
-	memcpy(&timbre->common, mem, 14);
-	unsigned int memPos = 14;
-	char drumname[11];
-	memset(drumname, 0, 11);
-	memcpy(drumname, timbre->common.name, 10);
-	for (int t = 0; t < 4; t++) {
-		if (((timbre->common.pmute >> t) & 0x1) == 0x1) {
-			if (memPos + 58 >= memLen) {
-				return false;
-			}
-			memcpy(&timbre->partial[t], mem + memPos, 58);
-			memPos += 58;
-		}
-	}
-	return true;
-}
-
-bool Synth::initRhythmTimbres(Bit16u mapAddress, Bit16u count) {
-	const Bit8u *drumMap = &controlROMData[mapAddress];
-	int timbreNum = 192;
-	for (Bit16u i = 0; i < count * 2; i += 2) {
-		Bit16u address = (drumMap[i + 1] << 8) | drumMap[i];
-		/*
-		// This check is nonsensical when the control ROM is the full 64KB addressable by 16-bit absolute pointers (which it is)
-		if (address >= CONTROL_ROM_SIZE) {
-			printDebug("Control ROM error: Timbre map entry 0x%04x points to invalid timbre address 0x%04x", i, address);
-			return false;
-		}
-		*/
-		if (!initRhythmTimbre(timbreNum++, &controlROMData[address], CONTROL_ROM_SIZE - address)) {
-			printDebug("Control ROM error: Timbre map entry 0x%04x points to invalid timbre 0x%04x", i, address);
-			return false;
-		}
-	}
-	return true;
-}
-
-bool Synth::initTimbres(Bit16u mapAddress, Bit16u offset, int startTimbre) {
-	for (Bit16u i = mapAddress; i < mapAddress + 0x80; i += 2) {
-		Bit16u address = (controlROMData[i + 1] << 8) | controlROMData[i];
-		if (address + sizeof(TimbreParam) > CONTROL_ROM_SIZE) {
-			printDebug("Control ROM error: Timbre map entry 0x%04x points to invalid timbre address 0x%04x", i, address);
-			return false;
-		}
-		address = address + offset;
-		TimbreParam *timbre = &mt32ram.timbres[startTimbre++].timbre;
-		memcpy(timbre, &controlROMData[address], sizeof(TimbreParam));
-	}
-	return true;
-}
-
-bool Synth::open(SynthProperties &useProp) {
-	if (isOpen)
-		return false;
-
-	myProp = useProp;
-	if (useProp.baseDir != NULL) {
-		myProp.baseDir = new char[strlen(useProp.baseDir) + 1];
-		strcpy(myProp.baseDir, useProp.baseDir);
-	}
-
-	// This is to help detect bugs
-	memset(&mt32ram, '?', sizeof(mt32ram));
-
-	printDebug("Loading Control ROM");
-	if (!loadControlROM("CM32L_CONTROL.ROM")) {
-		if (!loadControlROM("MT32_CONTROL.ROM")) {
-			printDebug("Init Error - Missing or invalid MT32_CONTROL.ROM");
-			report(ReportType_errorControlROM, NULL);
-			return false;
-		}
-	}
-
-	// 512KB PCM ROM for MT-32, etc.
-	// 1MB PCM ROM for CM-32L, LAPC-I, CM-64, CM-500
-	// Note that the size below is given in samples (16-bit), not bytes
-	pcmROMSize = controlROMMap->pcmCount == 256 ? 512 * 1024 : 256 * 1024;
-	pcmROMData = new Bit16s[pcmROMSize];
-
-	printDebug("Loading PCM ROM");
-	if (!loadPCMROM("CM32L_PCM.ROM")) {
-		if (!loadPCMROM("MT32_PCM.ROM")) {
-			printDebug("Init Error - Missing MT32_PCM.ROM");
-			report(ReportType_errorPCMROM, NULL);
-			return false;
-		}
-	}
-
-	printDebug("Initialising Timbre Bank A");
-	if (!initTimbres(controlROMMap->timbreAMap, controlROMMap->timbreAOffset, 0)) {
-		return false;
-	}
-
-	printDebug("Initialising Timbre Bank B");
-	if (!initTimbres(controlROMMap->timbreBMap, controlROMMap->timbreBOffset, 64)) {
-		return false;
-	}
-
-	printDebug("Initialising Timbre Bank R");
-	if (!initRhythmTimbres(controlROMMap->timbreRMap, controlROMMap->timbreRCount)) {
-		return false;
-	}
-
-	printDebug("Initialising Timbre Bank M");
-	// CM-64 seems to initialise all bytes in this bank to 0.
-	memset(&mt32ram.timbres[128], 0, sizeof (mt32ram.timbres[128]) * 64);
-
-	partialManager = new PartialManager(this);
-
-	pcmWaves = new PCMWaveEntry[controlROMMap->pcmCount];
-
-	printDebug("Initialising PCM List");
-	initPCMList(controlROMMap->pcmTable, controlROMMap->pcmCount);
-
-	printDebug("Initialising Rhythm Temp");
-	memcpy(mt32ram.rhythmSettings, &controlROMData[controlROMMap->rhythmSettings], controlROMMap->rhythmSettingsCount * 4);
-
-	printDebug("Initialising Patches");
-	for (Bit8u i = 0; i < 128; i++) {
-		PatchParam *patch = &mt32ram.patches[i];
-		patch->timbreGroup = i / 64;
-		patch->timbreNum = i % 64;
-		patch->keyShift = 24;
-		patch->fineTune = 50;
-		patch->benderRange = 12;
-		patch->assignMode = 0;
-		patch->reverbSwitch = 1;
-		patch->dummy = 0;
-	}
-
-	printDebug("Initialising System");
-	// The MT-32 manual claims that "Standard pitch" is 442Hz.
-	mt32ram.system.masterTune = 0x40; // Confirmed on CM-64 as 0x4A, but SCUMM games use 0x40 and we don't want to initialise twice
-	mt32ram.system.reverbMode = 0; // Confirmed
-	mt32ram.system.reverbTime = 5; // Confirmed
-	mt32ram.system.reverbLevel = 3; // Confirmed
-	memcpy(mt32ram.system.reserveSettings, &controlROMData[controlROMMap->reserveSettings], 9); // Confirmed
-	for (Bit8u i = 0; i < 9; i++) {
-		// This is the default: {1, 2, 3, 4, 5, 6, 7, 8, 9}
-		// An alternative configuration can be selected by holding "Master Volume"
-		// and pressing "PART button 1" on the real MT-32's frontpanel.
-		// The channel assignment is then {0, 1, 2, 3, 4, 5, 6, 7, 9}
-		mt32ram.system.chanAssign[i] = i + 1;
-	}
-	mt32ram.system.masterVol = 100; // Confirmed
-	if (!refreshSystem())
-		return false;
-
-	for (int i = 0; i < 8; i++) {
-		mt32ram.patchSettings[i].outlevel = 80;
-		mt32ram.patchSettings[i].panpot = controlROMData[controlROMMap->panSettings + i];
-		memset(mt32ram.patchSettings[i].dummyv, 0, sizeof(mt32ram.patchSettings[i].dummyv));
-		parts[i] = new Part(this, i);
-		parts[i]->setProgram(controlROMData[controlROMMap->programSettings + i]);
-	}
-	parts[8] = new RhythmPart(this, 8);
-
-	// For resetting mt32 mid-execution
-	mt32default = mt32ram;
-
-	iirFilter = &iir_filter_normal;
-
-#ifdef MT32EMU_HAVE_X86
-	bool availableSSE = DetectSIMD();
-	bool available3DNow = Detect3DNow();
-
-	if (availableSSE)
-		report(ReportType_availableSSE, NULL);
-	if (available3DNow)
-		report(ReportType_available3DNow, NULL);
-
-	if (available3DNow) {
-		printDebug("Detected and using SIMD (AMD 3DNow) extensions");
-		iirFilter = &iir_filter_3dnow;
-		report(ReportType_using3DNow, NULL);
-	} else if (availableSSE) {
-		printDebug("Detected and using SIMD (Intel SSE) extensions");
-		iirFilter = &iir_filter_sse;
-		report(ReportType_usingSSE, NULL);
-	}
-#endif
-
-	isOpen = true;
-	isEnabled = false;
-
-	printDebug("*** Initialisation complete ***");
-	return true;
-}
-
-void Synth::close(void) {
-	if (!isOpen)
-		return;
-
-	tables.freeNotes();
-	if (partialManager != NULL) {
-		delete partialManager;
-		partialManager = NULL;
-	}
-
-	if (reverbModel != NULL) {
-		delete reverbModel;
-		reverbModel = NULL;
-	}
-
-	for (int i = 0; i < 9; i++) {
-		if (parts[i] != NULL) {
-			delete parts[i];
-			parts[i] = NULL;
-		}
-	}
-	if (myProp.baseDir != NULL) {
-		delete myProp.baseDir;
-		myProp.baseDir = NULL;
-	}
-
-	delete[] pcmWaves;
-	delete[] pcmROMData;
-	isOpen = false;
-}
-
-void Synth::playMsg(Bit32u msg) {
-	// FIXME: Implement active sensing
-	unsigned char code     = (unsigned char)((msg & 0x0000F0) >> 4);
-	unsigned char chan     = (unsigned char) (msg & 0x00000F);
-	unsigned char note     = (unsigned char)((msg & 0x00FF00) >> 8);
-	unsigned char velocity = (unsigned char)((msg & 0xFF0000) >> 16);
-	isEnabled = true;
-
-	//printDebug("Playing chan %d, code 0x%01x note: 0x%02x", chan, code, note);
-
-	signed char part = chantable[chan];
-	if (part < 0 || part > 8) {
-		printDebug("Play msg on unreg chan %d (%d): code=0x%01x, vel=%d", chan, part, code, velocity);
-		return;
-	}
-	playMsgOnPart(part, code, note, velocity);
-}
-
-void Synth::playMsgOnPart(unsigned char part, unsigned char code, unsigned char note, unsigned char velocity) {
-	Bit32u bend;
-
-	//printDebug("Synth::playMsg(0x%02x)",msg);
-	switch (code) {
-	case 0x8:
-		//printDebug("Note OFF - Part %d", part);
-		// The MT-32 ignores velocity for note off
-		parts[part]->stopNote(note);
-		break;
-	case 0x9:
-		//printDebug("Note ON - Part %d, Note %d Vel %d", part, note, velocity);
-		if (velocity == 0) {
-			// MIDI defines note-on with velocity 0 as being the same as note-off with velocity 40
-			parts[part]->stopNote(note);
-		} else {
-			parts[part]->playNote(note, velocity);
-		}
-		break;
-	case 0xB: // Control change
-		switch (note) {
-		case 0x01:  // Modulation
-			//printDebug("Modulation: %d", velocity);
-			parts[part]->setModulation(velocity);
-			break;
-		case 0x07:  // Set volume
-			//printDebug("Volume set: %d", velocity);
-			parts[part]->setVolume(velocity);
-			break;
-		case 0x0A:  // Pan
-			//printDebug("Pan set: %d", velocity);
-			parts[part]->setPan(velocity);
-			break;
-		case 0x0B:
-			//printDebug("Expression set: %d", velocity);
-			parts[part]->setExpression(velocity);
-			break;
-		case 0x40: // Hold (sustain) pedal
-			//printDebug("Hold pedal set: %d", velocity);
-			parts[part]->setHoldPedal(velocity>=64);
-			break;
-
-		case 0x79: // Reset all controllers
-			//printDebug("Reset all controllers");
-			//FIXME: Check for accuracy against real thing
-			parts[part]->setVolume(100);
-			parts[part]->setExpression(127);
-			parts[part]->setPan(64);
-			parts[part]->setBend(0x2000);
-			parts[part]->setHoldPedal(false);
-			break;
-
-		case 0x7B: // All notes off
-			//printDebug("All notes off");
-			parts[part]->allNotesOff();
-			break;
-
-		default:
-			printDebug("Unknown MIDI Control code: 0x%02x - vel 0x%02x", note, velocity);
-			break;
-		}
-
-		break;
-	case 0xC: // Program change
-		//printDebug("Program change %01x", note);
-		parts[part]->setProgram(note);
-		break;
-	case 0xE: // Pitch bender
-		bend = (velocity << 7) | (note);
-		//printDebug("Pitch bender %02x", bend);
-		parts[part]->setBend(bend);
-		break;
-	default:
-		printDebug("Unknown Midi code: 0x%01x - %02x - %02x", code, note, velocity);
-		break;
-	}
-
-	//midiOutShortMsg(m_out, msg);
-}
-
-void Synth::playSysex(const Bit8u *sysex, Bit32u len) {
-	if (len < 2) {
-		printDebug("playSysex: Message is too short for sysex (%d bytes)", len);
-	}
-	if (sysex[0] != 0xF0) {
-		printDebug("playSysex: Message lacks start-of-sysex (0xF0)");
-		return;
-	}
-	// Due to some programs (e.g. Java) sending buffers with junk at the end, we have to go through and find the end marker rather than relying on len.
-	Bit32u endPos;
-	for (endPos = 1; endPos < len; endPos++)
-	{
-		if (sysex[endPos] == 0xF7)
-			break;
-	}
-	if (endPos == len) {
-		printDebug("playSysex: Message lacks end-of-sysex (0xf7)");
-		return;
-	}
-	playSysexWithoutFraming(sysex + 1, endPos - 1);
-}
-
-void Synth::playSysexWithoutFraming(const Bit8u *sysex, Bit32u len) {
-	if (len < 4) {
-		printDebug("playSysexWithoutFraming: Message is too short (%d bytes)!", len);
-		return;
-	}
-	if (sysex[0] != SYSEX_MANUFACTURER_ROLAND) {
-		printDebug("playSysexWithoutFraming: Header not intended for this device manufacturer: %02x %02x %02x %02x", (int)sysex[0], (int)sysex[1], (int)sysex[2], (int)sysex[3]);
-		return;
-	}
-	if (sysex[2] == SYSEX_MDL_D50) {
-		printDebug("playSysexWithoutFraming: Header is intended for model D-50 (not yet supported): %02x %02x %02x %02x", (int)sysex[0], (int)sysex[1], (int)sysex[2], (int)sysex[3]);
-		return;
-	}
-	else if (sysex[2] != SYSEX_MDL_MT32) {
-		printDebug("playSysexWithoutFraming: Header not intended for model MT-32: %02x %02x %02x %02x", (int)sysex[0], (int)sysex[1], (int)sysex[2], (int)sysex[3]);
-		return;
-	}
-	playSysexWithoutHeader(sysex[1], sysex[3], sysex + 4, len - 4);
-}
-
-void Synth::playSysexWithoutHeader(unsigned char device, unsigned char command, const Bit8u *sysex, Bit32u len) {
-	if (device > 0x10) {
-		// We have device ID 0x10 (default, but changeable, on real MT-32), < 0x10 is for channels
-		printDebug("playSysexWithoutHeader: Message is not intended for this device ID (provided: %02x, expected: 0x10 or channel)", (int)device);
-		return;
-	}
-	if (len < 4) {
-		printDebug("playSysexWithoutHeader: Message is too short (%d bytes)!", len);
-		return;
-	}
-	unsigned char checksum = calcSysexChecksum(sysex, len - 1, 0);
-	if (checksum != sysex[len - 1]) {
-		printDebug("playSysexWithoutHeader: Message checksum is incorrect (provided: %02x, expected: %02x)!", sysex[len - 1], checksum);
-		return;
-	}
-	len -= 1; // Exclude checksum
-	switch (command) {
-	case SYSEX_CMD_DT1:
-		writeSysex(device, sysex, len);
-		break;
-	case SYSEX_CMD_RQ1:
-		readSysex(device, sysex, len);
-		break;
-	default:
-		printDebug("playSysexWithoutFraming: Unsupported command %02x", command);
-		return;
-	}
-}
-
-void Synth::readSysex(unsigned char /*device*/, const Bit8u * /*sysex*/, Bit32u /*len*/) {
-}
-
-const MemoryRegion memoryRegions[8] = {
-	{MR_PatchTemp,  MT32EMU_MEMADDR(0x030000), sizeof(MemParams::PatchTemp), 9},
-	{MR_RhythmTemp, MT32EMU_MEMADDR(0x030110), sizeof(MemParams::RhythmTemp), 85},
-	{MR_TimbreTemp, MT32EMU_MEMADDR(0x040000), sizeof(TimbreParam), 8},
-	{MR_Patches,    MT32EMU_MEMADDR(0x050000), sizeof(PatchParam), 128},
-	{MR_Timbres,    MT32EMU_MEMADDR(0x080000), sizeof(MemParams::PaddedTimbre), 64 + 64 + 64 + 64},
-	{MR_System,     MT32EMU_MEMADDR(0x100000), sizeof(MemParams::SystemArea), 1},
-	{MR_Display,    MT32EMU_MEMADDR(0x200000), MAX_SYSEX_SIZE - 1, 1},
-	{MR_Reset,      MT32EMU_MEMADDR(0x7F0000), 0x3FFF, 1}
-};
-
-const int NUM_REGIONS = sizeof(memoryRegions) / sizeof(MemoryRegion);
-
-void Synth::writeSysex(unsigned char device, const Bit8u *sysex, Bit32u len) {
-	Bit32u addr = (sysex[0] << 16) | (sysex[1] << 8) | (sysex[2]);
-	addr = MT32EMU_MEMADDR(addr);
-	sysex += 3;
-	len -= 3;
-	//printDebug("Sysex addr: 0x%06x", MT32EMU_SYSEXMEMADDR(addr));
-	// NOTE: Please keep both lower and upper bounds in each check, for ease of reading
-
-	// Process channel-specific sysex by converting it to device-global
-	if (device < 0x10) {
-		printDebug("WRITE-CHANNEL: Channel %d temp area 0x%06x", device, MT32EMU_SYSEXMEMADDR(addr));
-		if (/*addr >= MT32EMU_MEMADDR(0x000000) && */addr < MT32EMU_MEMADDR(0x010000)) {
-			int offset;
-			if (chantable[device] == -1) {
-				printDebug(" (Channel not mapped to a partial... 0 offset)");
-				offset = 0;
-			} else if (chantable[device] == 8) {
-				printDebug(" (Channel mapped to rhythm... 0 offset)");
-				offset = 0;
-			} else {
-				offset = chantable[device] * sizeof(MemParams::PatchTemp);
-				printDebug(" (Setting extra offset to %d)", offset);
-			}
-			addr += MT32EMU_MEMADDR(0x030000) + offset;
-		} else if (/*addr >= 0x010000 && */ addr < MT32EMU_MEMADDR(0x020000)) {
-			addr += MT32EMU_MEMADDR(0x030110) - MT32EMU_MEMADDR(0x010000);
-		} else if (/*addr >= 0x020000 && */ addr < MT32EMU_MEMADDR(0x030000)) {
-			int offset;
-			if (chantable[device] == -1) {
-				printDebug(" (Channel not mapped to a partial... 0 offset)");
-				offset = 0;
-			} else if (chantable[device] == 8) {
-				printDebug(" (Channel mapped to rhythm... 0 offset)");
-				offset = 0;
-			} else {
-				offset = chantable[device] * sizeof(TimbreParam);
-				printDebug(" (Setting extra offset to %d)", offset);
-			}
-			addr += MT32EMU_MEMADDR(0x040000) - MT32EMU_MEMADDR(0x020000) + offset;
-		} else {
-			printDebug("PlaySysexWithoutHeader: Invalid channel %d address 0x%06x", device, MT32EMU_SYSEXMEMADDR(addr));
-			return;
-		}
-	}
-
-	// Process device-global sysex (possibly converted from channel-specific sysex above)
-	for (;;) {
-		// Find the appropriate memory region
-		int regionNum;
-		const MemoryRegion *region = NULL; // Initialised to please compiler
-		for (regionNum = 0; regionNum < NUM_REGIONS; regionNum++) {
-			region = &memoryRegions[regionNum];
-			if (region->contains(addr)) {
-				writeMemoryRegion(region, addr, region->getClampedLen(addr, len), sysex);
-				break;
-			}
-		}
-		if (regionNum == NUM_REGIONS) {
-			printDebug("Sysex write to unrecognised address %06x, len %d", MT32EMU_SYSEXMEMADDR(addr), len);
-			break;
-		}
-		Bit32u next = region->next(addr, len);
-		if (next == 0) {
-			break;
-		}
-		addr += next;
-		sysex += next;
-		len -= next;
-	}
-}
-
-void Synth::readMemory(Bit32u addr, Bit32u len, Bit8u *data) {
-	int regionNum;
-	const MemoryRegion *region = NULL;
-	for (regionNum = 0; regionNum < NUM_REGIONS; regionNum++) {
-		region = &memoryRegions[regionNum];
-		if (region->contains(addr)) {
-			readMemoryRegion(region, addr, len, data);
-			break;
-		}
-	}
-}
-
-void Synth::readMemoryRegion(const MemoryRegion *region, Bit32u addr, Bit32u len, Bit8u *data) {
-	unsigned int first = region->firstTouched(addr);
-	//unsigned int last = region->lastTouched(addr, len);
-	unsigned int off = region->firstTouchedOffset(addr);
-	len = region->getClampedLen(addr, len);
-
-	unsigned int m;
-
-	switch (region->type) {
-	case MR_PatchTemp:
-		for (m = 0; m < len; m++)
-			data[m] = ((Bit8u *)&mt32ram.patchSettings[first])[off + m];
-		break;
-	case MR_RhythmTemp:
-		for (m = 0; m < len; m++)
-			data[m] = ((Bit8u *)&mt32ram.rhythmSettings[first])[off + m];
-		break;
-	case MR_TimbreTemp:
-		for (m = 0; m < len; m++)
-			data[m] = ((Bit8u *)&mt32ram.timbreSettings[first])[off + m];
-		break;
-	case MR_Patches:
-		for (m = 0; m < len; m++)
-			data[m] = ((Bit8u *)&mt32ram.patches[first])[off + m];
-		break;
-	case MR_Timbres:
-		for (m = 0; m < len; m++)
-			data[m] = ((Bit8u *)&mt32ram.timbres[first])[off + m];
-		break;
-	case MR_System:
-		for (m = 0; m < len; m++)
-			data[m] = ((Bit8u *)&mt32ram.system)[m + off];
-		break;
-	default:
-		for (m = 0; m < len; m += 2) {
-			data[m] = 0xff;
-			if (m + 1 < len) {
-				data[m+1] = (Bit8u)region->type;
-			}
-		}
-		// TODO: Don't care about the others ATM
-		break;
-	}
-
-}
-
-void Synth::writeMemoryRegion(const MemoryRegion *region, Bit32u addr, Bit32u len, const Bit8u *data) {
-	unsigned int first = region->firstTouched(addr);
-	unsigned int last = region->lastTouched(addr, len);
-	unsigned int off = region->firstTouchedOffset(addr);
-	switch (region->type) {
-	case MR_PatchTemp:
-		for (unsigned int m = 0; m < len; m++) {
-			((Bit8u *)&mt32ram.patchSettings[first])[off + m] = data[m];
-		}
-		//printDebug("Patch temp: Patch %d, offset %x, len %d", off/16, off % 16, len);
-
-		for (unsigned int i = first; i <= last; i++) {
-			int absTimbreNum = mt32ram.patchSettings[i].patch.timbreGroup * 64 + mt32ram.patchSettings[i].patch.timbreNum;
-			char timbreName[11];
-			memcpy(timbreName, mt32ram.timbres[absTimbreNum].timbre.common.name, 10);
-			timbreName[10] = 0;
-			printDebug("WRITE-PARTPATCH (%d-%d@%d..%d): %d; timbre=%d (%s), outlevel=%d", first, last, off, off + len, i, absTimbreNum, timbreName, mt32ram.patchSettings[i].outlevel);
-			if (parts[i] != NULL) {
-				if (i != 8) {
-					// Note: Confirmed on CM-64 that we definitely *should* update the timbre here,
-					// but only in the case that the sysex actually writes to those values
-					if (i == first && off > 2) {
-						printDebug(" (Not updating timbre, since those values weren't touched)");
-					} else {
-						parts[i]->setTimbre(&mt32ram.timbres[parts[i]->getAbsTimbreNum()].timbre);
-					}
-				}
-				parts[i]->refresh();
-			}
-		}
-		break;
-	case MR_RhythmTemp:
-		for (unsigned int m = 0; m < len; m++)
-			((Bit8u *)&mt32ram.rhythmSettings[first])[off + m] = data[m];
-		for (unsigned int i = first; i <= last; i++) {
-			int timbreNum = mt32ram.rhythmSettings[i].timbre;
-			char timbreName[11];
-			if (timbreNum < 94) {
-				memcpy(timbreName, mt32ram.timbres[128 + timbreNum].timbre.common.name, 10);
-				timbreName[10] = 0;
-			} else {
-				strcpy(timbreName, "[None]");
-			}
-			printDebug("WRITE-RHYTHM (%d-%d@%d..%d): %d; level=%02x, panpot=%02x, reverb=%02x, timbre=%d (%s)", first, last, off, off + len, i, mt32ram.rhythmSettings[i].outlevel, mt32ram.rhythmSettings[i].panpot, mt32ram.rhythmSettings[i].reverbSwitch, mt32ram.rhythmSettings[i].timbre, timbreName);
-		}
-		if (parts[8] != NULL) {
-			parts[8]->refresh();
-		}
-		break;
-	case MR_TimbreTemp:
-		for (unsigned int m = 0; m < len; m++)
-			((Bit8u *)&mt32ram.timbreSettings[first])[off + m] = data[m];
-		for (unsigned int i = first; i <= last; i++) {
-			char instrumentName[11];
-			memcpy(instrumentName, mt32ram.timbreSettings[i].common.name, 10);
-			instrumentName[10] = 0;
-			printDebug("WRITE-PARTTIMBRE (%d-%d@%d..%d): timbre=%d (%s)", first, last, off, off + len, i, instrumentName);
-			if (parts[i] != NULL) {
-				parts[i]->refresh();
-			}
-		}
-		break;
-	case MR_Patches:
-		for (unsigned int m = 0; m < len; m++)
-			((Bit8u *)&mt32ram.patches[first])[off + m] = data[m];
-		for (unsigned int i = first; i <= last; i++) {
-			PatchParam *patch = &mt32ram.patches[i];
-			int patchAbsTimbreNum = patch->timbreGroup * 64 + patch->timbreNum;
-			char instrumentName[11];
-			memcpy(instrumentName, mt32ram.timbres[patchAbsTimbreNum].timbre.common.name, 10);
-			instrumentName[10] = 0;
-			Bit8u *n = (Bit8u *)patch;
-			printDebug("WRITE-PATCH (%d-%d@%d..%d): %d; timbre=%d (%s) %02X%02X%02X%02X%02X%02X%02X%02X", first, last, off, off + len, i, patchAbsTimbreNum, instrumentName, n[0], n[1], n[2], n[3], n[4], n[5], n[6], n[7]);
-			// FIXME:KG: The below is definitely dodgy. We just guess that this is the patch that the part was using
-			// based on a timbre match (but many patches could have the same timbre!)
-			// If this refresh is really correct, we should store the patch number in use by each part.
-			/*
-			for (int part = 0; part < 8; part++) {
-				if (parts[part] != NULL) {
-					int partPatchAbsTimbreNum = mt32ram.patchSettings[part].patch.timbreGroup * 64 + mt32ram.patchSettings[part].patch.timbreNum;
-					if (parts[part]->getAbsTimbreNum() == patchAbsTimbreNum) {
-						parts[part]->setPatch(patch);
-						parts[part]->RefreshPatch();
-					}
-				}
-			}
-			*/
-		}
-		break;
-	case MR_Timbres:
-		// Timbres
-		first += 128;
-		last += 128;
-		for (unsigned int m = 0; m < len; m++)
-			((Bit8u *)&mt32ram.timbres[first])[off + m] = data[m];
-		for (unsigned int i = first; i <= last; i++) {
-			char instrumentName[11];
-			memcpy(instrumentName, mt32ram.timbres[i].timbre.common.name, 10);
-			instrumentName[10] = 0;
-			printDebug("WRITE-TIMBRE (%d-%d@%d..%d): %d; name=\"%s\"", first, last, off, off + len, i, instrumentName);
-			// FIXME:KG: Not sure if the stuff below should be done (for rhythm and/or parts)...
-			// Does the real MT-32 automatically do this?
-			for (unsigned int part = 0; part < 9; part++) {
-				if (parts[part] != NULL) {
-					parts[part]->refreshTimbre(i);
-				}
-			}
-		}
-		break;
-	case MR_System:
-		for (unsigned int m = 0; m < len; m++)
-			((Bit8u *)&mt32ram.system)[m + off] = data[m];
-
-		report(ReportType_devReconfig, NULL);
-
-		printDebug("WRITE-SYSTEM:");
-		refreshSystem();
-		break;
-	case MR_Display:
-		char buf[MAX_SYSEX_SIZE];
-		memcpy(&buf, &data[0], len);
-		buf[len] = 0;
-		printDebug("WRITE-LCD: %s", buf);
-		report(ReportType_lcdMessage, buf);
-		break;
-	case MR_Reset:
-		printDebug("RESET");
-		report(ReportType_devReset, NULL);
-		partialManager->deactivateAll();
-		mt32ram = mt32default;
-		for (int i = 0; i < 9; i++) {
-			parts[i]->refresh();
-		}
-		isEnabled = false;
-		break;
-	}
-}
-
-bool Synth::refreshSystem() {
-	memset(chantable, -1, sizeof(chantable));
-
-	for (unsigned int i = 0; i < 9; i++) {
-		//LOG(LOG_MISC|LOG_ERROR,"Part %d set to MIDI channel %d",i,mt32ram.system.chanAssign[i]);
-		if (mt32ram.system.chanAssign[i] == 16 && parts[i] != NULL) {
-			parts[i]->allSoundOff();
-		} else {
-			chantable[(int)mt32ram.system.chanAssign[i]] = (char)i;
-		}
-	}
-	//FIXME:KG: This is just an educated guess.
-	// The LAPC-I documentation claims a range of 427.5Hz-452.6Hz (similar to what we have here)
-	// The MT-32 documentation claims a range of 432.1Hz-457.6Hz
-	masterTune = 440.0f * powf(2.0f, (mt32ram.system.masterTune - 64.0f) / (128.0f * 12.0f));
-	printDebug(" Master Tune: %f", (double)masterTune);
-	printDebug(" Reverb: mode=%d, time=%d, level=%d", mt32ram.system.reverbMode, mt32ram.system.reverbTime, mt32ram.system.reverbLevel);
-	report(ReportType_newReverbMode,  &mt32ram.system.reverbMode);
-	report(ReportType_newReverbTime,  &mt32ram.system.reverbTime);
-	report(ReportType_newReverbLevel, &mt32ram.system.reverbLevel);
-
-	if (myProp.useDefaultReverb) {
-		initReverb(mt32ram.system.reverbMode, mt32ram.system.reverbTime, mt32ram.system.reverbLevel);
-	} else {
-		initReverb(myProp.reverbType, myProp.reverbTime, mt32ram.system.reverbLevel);
-	}
-
-	Bit8u *rset = mt32ram.system.reserveSettings;
-	printDebug(" Partial reserve: 1=%02d 2=%02d 3=%02d 4=%02d 5=%02d 6=%02d 7=%02d 8=%02d Rhythm=%02d", rset[0], rset[1], rset[2], rset[3], rset[4], rset[5], rset[6], rset[7], rset[8]);
-	int pr = partialManager->setReserve(rset);
-	if (pr != 32)
-		printDebug(" (Partial Reserve Table with less than 32 partials reserved!)");
-	rset = mt32ram.system.chanAssign;
-	printDebug(" Part assign:     1=%02d 2=%02d 3=%02d 4=%02d 5=%02d 6=%02d 7=%02d 8=%02d Rhythm=%02d", rset[0], rset[1], rset[2], rset[3], rset[4], rset[5], rset[6], rset[7], rset[8]);
-	printDebug(" Master volume: %d", mt32ram.system.masterVol);
-	masterVolume = (Bit16u)(mt32ram.system.masterVol * 32767 / 100);
-	if (!tables.init(this, pcmWaves, (float)myProp.sampleRate, masterTune)) {
-		report(ReportType_errorSampleRate, NULL);
-		return false;
-	}
-	return true;
-}
-
-bool Synth::dumpTimbre(File *file, const TimbreParam *timbre, Bit32u address) {
-	// Sysex header
-	if (!file->writeBit8u(0xF0))
-		return false;
-	if (!file->writeBit8u(0x41))
-		return false;
-	if (!file->writeBit8u(0x10))
-		return false;
-	if (!file->writeBit8u(0x16))
-		return false;
-	if (!file->writeBit8u(0x12))
-		return false;
-
-	char lsb = (char)(address & 0x7f);
-	char isb = (char)((address >> 7) & 0x7f);
-	char msb = (char)(((address >> 14) & 0x7f) | 0x08);
-
-	//Address
-	if (!file->writeBit8u(msb))
-		return false;
-	if (!file->writeBit8u(isb))
-		return false;
-	if (!file->writeBit8u(lsb))
-		return false;
-
-	//Data
-	if (file->write(timbre, 246) != 246)
-		return false;
-
-	//Checksum
-	unsigned char checksum = calcSysexChecksum((const Bit8u *)timbre, 246, msb + isb + lsb);
-	if (!file->writeBit8u(checksum))
-		return false;
-
-	//End of sysex
-	if (!file->writeBit8u(0xF7))
-		return false;
-	return true;
-}
-
-int Synth::dumpTimbres(const char *filename, int start, int len) {
-	File *file = openFile(filename, File::OpenMode_write);
-	if (file == NULL)
-		return -1;
-
-	for (int timbreNum = start; timbreNum < start + len; timbreNum++) {
-		int useaddr = (timbreNum - start) * 256;
-		TimbreParam *timbre = &mt32ram.timbres[timbreNum].timbre;
-		if (!dumpTimbre(file, timbre, useaddr))
-			break;
-	}
-	closeFile(file);
-	return 0;
-}
-
-void ProduceOutput1(Bit16s *useBuf, Bit16s *stream, Bit32u len, Bit16s volume) {
-#if MT32EMU_USE_MMX > 2
-	//FIXME:KG: This appears to introduce crackle
-	int donelen = i386_produceOutput1(useBuf, stream, len, volume);
-	len -= donelen;
-	stream += donelen * 2;
-	useBuf += donelen * 2;
-#endif
-	int end = len * 2;
-	while (end--) {
-		*stream = *stream + (Bit16s)(((Bit32s)*useBuf++ * (Bit32s)volume)>>15);
-		stream++;
-	}
-}
-
-void Synth::render(Bit16s *stream, Bit32u len) {
-	memset(stream, 0, len * sizeof (Bit16s) * 2);
-	if (!isEnabled)
-		return;
-	while (len > 0) {
-		Bit32u thisLen = len > MAX_SAMPLE_OUTPUT ? MAX_SAMPLE_OUTPUT : len;
-		doRender(stream, thisLen);
-		len -= thisLen;
-		stream += 2 * thisLen;
-	}
-}
-
-void Synth::doRender(Bit16s *stream, Bit32u len) {
-	partialManager->ageAll();
-
-	if (myProp.useReverb) {
-		for (unsigned int i = 0; i < MT32EMU_MAX_PARTIALS; i++) {
-			if (partialManager->shouldReverb(i)) {
-				if (partialManager->produceOutput(i, &tmpBuffer[0], len)) {
-					ProduceOutput1(&tmpBuffer[0], stream, len, masterVolume);
-				}
-			}
-		}
-		Bit32u m = 0;
-		for (unsigned int i = 0; i < len; i++) {
-			sndbufl[i] = (float)stream[m] / 32767.0f;
-			m++;
-			sndbufr[i] = (float)stream[m] / 32767.0f;
-			m++;
-		}
-		reverbModel->processreplace(sndbufl, sndbufr, outbufl, outbufr, len, 1);
-		m=0;
-		for (unsigned int i = 0; i < len; i++) {
-			stream[m] = (Bit16s)(outbufl[i] * 32767.0f);
-			m++;
-			stream[m] = (Bit16s)(outbufr[i] * 32767.0f);
-			m++;
-		}
-		for (unsigned int i = 0; i < MT32EMU_MAX_PARTIALS; i++) {
-			if (!partialManager->shouldReverb(i)) {
-				if (partialManager->produceOutput(i, &tmpBuffer[0], len)) {
-					ProduceOutput1(&tmpBuffer[0], stream, len, masterVolume);
-				}
-			}
-		}
-	} else {
-		for (unsigned int i = 0; i < MT32EMU_MAX_PARTIALS; i++) {
-			if (partialManager->produceOutput(i, &tmpBuffer[0], len))
-				ProduceOutput1(&tmpBuffer[0], stream, len, masterVolume);
-		}
-	}
-
-	partialManager->clearAlreadyOutputed();
-
-#if MT32EMU_MONITOR_PARTIALS == 1
-	samplepos += len;
-	if (samplepos > myProp.SampleRate * 5) {
-		samplepos = 0;
-		int partialUsage[9];
-		partialManager->GetPerPartPartialUsage(partialUsage);
-		printDebug("1:%02d 2:%02d 3:%02d 4:%02d 5:%02d 6:%02d 7:%02d 8:%02d", partialUsage[0], partialUsage[1], partialUsage[2], partialUsage[3], partialUsage[4], partialUsage[5], partialUsage[6], partialUsage[7]);
-		printDebug("Rhythm: %02d  TOTAL: %02d", partialUsage[8], MT32EMU_MAX_PARTIALS - partialManager->GetFreePartialCount());
-	}
-#endif
-}
-
-const Partial *Synth::getPartial(unsigned int partialNum) const {
-	return partialManager->getPartial(partialNum);
-}
-
-const Part *Synth::getPart(unsigned int partNum) const {
-	if (partNum > 8)
-		return NULL;
-	return parts[partNum];
-}
-
-}
diff --git a/sound/softsynth/mt32/synth.h b/sound/softsynth/mt32/synth.h
deleted file mode 100644
index 3fc303d322..0000000000
--- a/sound/softsynth/mt32/synth.h
+++ /dev/null
@@ -1,300 +0,0 @@
-/* Copyright (c) 2003-2005 Various contributors
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to
- * deal in the Software without restriction, including without limitation the
- * rights to use, copy, modify, merge, publish, distribute, sublicense, and/or
- * sell copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
- * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
- * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS
- * IN THE SOFTWARE.
- */
-
-#ifndef MT32EMU_SYNTH_H
-#define MT32EMU_SYNTH_H
-
-#include "common/scummsys.h"
-
-class revmodel;
-
-namespace MT32Emu {
-
-class File;
-class TableInitialiser;
-class Partial;
-class PartialManager;
-class Part;
-
-enum ReportType {
-	// Errors
-	ReportType_errorControlROM = 1,
-	ReportType_errorPCMROM,
-	ReportType_errorSampleRate,
-
-	// Progress
-	ReportType_progressInit,
-
-	// HW spec
-	ReportType_availableSSE,
-	ReportType_available3DNow,
-	ReportType_usingSSE,
-	ReportType_using3DNow,
-
-	// General info
-	ReportType_lcdMessage,
-	ReportType_devReset,
-	ReportType_devReconfig,
-	ReportType_newReverbMode,
-	ReportType_newReverbTime,
-	ReportType_newReverbLevel
-};
-
-struct SynthProperties {
-	// Sample rate to use in mixing
-	int sampleRate;
-
-	// Flag to activate reverb.  True = use reverb, False = no reverb
-	bool useReverb;
-	// True to use software set reverb settings, False to set reverb settings in
-	// following parameters
-	bool useDefaultReverb;
-	// When not using the default settings, this specifies one of the 4 reverb types
-	// 1 = Room 2 = Hall 3 = Plate 4 = Tap
-	unsigned char reverbType;
-	// This specifies the delay time, from 0-7 (not sure of the actual MT-32's measurement)
-	unsigned char reverbTime;
-	// This specifies the reverb level, from 0-7 (not sure of the actual MT-32's measurement)
-	unsigned char reverbLevel;
-	// The name of the directory in which the ROM and data files are stored (with trailing slash/backslash)
-	// Not used if "openFile" is set. May be NULL in any case.
-	char *baseDir;
-	// This is used as the first argument to all callbacks
-	void *userData;
-	// Callback for reporting various errors and information. May be NULL
-	int (*report)(void *userData, ReportType type, const void *reportData);
-	// Callback for debug messages, in vprintf() format
-	void (*printDebug)(void *userData, const char *fmt, va_list list);
-	// Callback for providing an implementation of File, opened and ready for use
-	// May be NULL, in which case a default implementation will be used.
-	File *(*openFile)(void *userData, const char *filename, File::OpenMode mode);
-	// Callback for closing a File. May be NULL, in which case the File will automatically be close()d/deleted.
-	void (*closeFile)(void *userData, File *file);
-};
-
-// This is the specification of the Callback routine used when calling the RecalcWaveforms
-// function
-typedef void (*recalcStatusCallback)(int percDone);
-
-// This external function recreates the base waveform file (waveforms.raw) using a specifed
-// sampling rate.  The callback routine provides interactivity to let the user know what
-// percentage is complete in regenerating the waveforms.  When a NULL pointer is used as the
-// callback routine, no status is reported.
-bool RecalcWaveforms(char * baseDir, int sampRate, recalcStatusCallback callBack);
-
-typedef float (*iir_filter_type)(float input,float *hist1_ptr, float *coef_ptr);
-
-const Bit8u SYSEX_MANUFACTURER_ROLAND = 0x41;
-
-const Bit8u SYSEX_MDL_MT32 = 0x16;
-const Bit8u SYSEX_MDL_D50 = 0x14;
-
-const Bit8u SYSEX_CMD_RQ1 = 0x11; // Request data #1
-const Bit8u SYSEX_CMD_DT1 = 0x12; // Data set 1
-const Bit8u SYSEX_CMD_WSD = 0x40; // Want to send data
-const Bit8u SYSEX_CMD_RQD = 0x41; // Request data
-const Bit8u SYSEX_CMD_DAT = 0x42; // Data set
-const Bit8u SYSEX_CMD_ACK = 0x43; // Acknowledge
-const Bit8u SYSEX_CMD_EOD = 0x45; // End of data
-const Bit8u SYSEX_CMD_ERR = 0x4E; // Communications error
-const Bit8u SYSEX_CMD_RJC = 0x4F; // Rejection
-
-const unsigned int CONTROL_ROM_SIZE = 64 * 1024;
-
-struct ControlROMPCMStruct
-{
-	Bit8u pos;
-	Bit8u len;
-	Bit8u pitchLSB;
-	Bit8u pitchMSB;
-};
-
-struct ControlROMMap {
-	Bit16u idPos;
-	Bit16u idLen;
-	const char *idBytes;
-	Bit16u pcmTable;
-	Bit16u pcmCount;
-	Bit16u timbreAMap;
-	Bit16u timbreAOffset;
-	Bit16u timbreBMap;
-	Bit16u timbreBOffset;
-	Bit16u timbreRMap;
-	Bit16u timbreRCount;
-	Bit16u rhythmSettings;
-	Bit16u rhythmSettingsCount;
-	Bit16u reserveSettings;
-	Bit16u panSettings;
-	Bit16u programSettings;
-};
-
-enum MemoryRegionType {
-	MR_PatchTemp, MR_RhythmTemp, MR_TimbreTemp, MR_Patches, MR_Timbres, MR_System, MR_Display, MR_Reset
-};
-
-class MemoryRegion {
-public:
-	MemoryRegionType type;
-	Bit32u startAddr, entrySize, entries;
-
-	int lastTouched(Bit32u addr, Bit32u len) const {
-		return (offset(addr) + len - 1) / entrySize;
-	}
-	int firstTouchedOffset(Bit32u addr) const {
-		return offset(addr) % entrySize;
-	}
-	int firstTouched(Bit32u addr) const {
-		return offset(addr) / entrySize;
-	}
-	Bit32u regionEnd() const {
-		return startAddr + entrySize * entries;
-	}
-	bool contains(Bit32u addr) const {
-		return addr >= startAddr && addr < regionEnd();
-	}
-	int offset(Bit32u addr) const {
-		return addr - startAddr;
-	}
-	Bit32u getClampedLen(Bit32u addr, Bit32u len) const {
-		if (addr + len > regionEnd())
-			return regionEnd() - addr;
-		return len;
-	}
-	Bit32u next(Bit32u addr, Bit32u len) const {
-		if (addr + len > regionEnd()) {
-			return regionEnd() - addr;
-		}
-		return 0;
-	}
-};
-
-
-class Synth {
-friend class Part;
-friend class RhythmPart;
-friend class Partial;
-friend class Tables;
-private:
-	bool isEnabled;
-
-	iir_filter_type iirFilter;
-
-	PCMWaveEntry *pcmWaves; // Array
-
-	const ControlROMMap *controlROMMap;
-	Bit8u controlROMData[CONTROL_ROM_SIZE];
-	Bit16s *pcmROMData;
-	int pcmROMSize; // This is in 16-bit samples, therefore half the number of bytes in the ROM
-
-	Bit8s chantable[32];
-
-	#if MT32EMU_MONITOR_PARTIALS == 1
-	static Bit32s samplepos = 0;
-	#endif
-
-	Tables tables;
-
-	MemParams mt32ram, mt32default;
-
-	revmodel *reverbModel;
-
-	float masterTune;
-	Bit16u masterVolume;
-
-	bool isOpen;
-
-	PartialManager *partialManager;
-	Part *parts[9];
-
-	Bit16s tmpBuffer[MAX_SAMPLE_OUTPUT * 2];
-	float sndbufl[MAX_SAMPLE_OUTPUT];
-	float sndbufr[MAX_SAMPLE_OUTPUT];
-	float outbufl[MAX_SAMPLE_OUTPUT];
-	float outbufr[MAX_SAMPLE_OUTPUT];
-
-	SynthProperties myProp;
-
-	bool loadPreset(File *file);
-	void initReverb(Bit8u newRevMode, Bit8u newRevTime, Bit8u newRevLevel);
-	void doRender(Bit16s * stream, Bit32u len);
-
-	void playAddressedSysex(unsigned char channel, const Bit8u *sysex, Bit32u len);
-	void readSysex(unsigned char channel, const Bit8u *sysex, Bit32u len);
-	void writeMemoryRegion(const MemoryRegion *region, Bit32u addr, Bit32u len, const Bit8u *data);
-	void readMemoryRegion(const MemoryRegion *region, Bit32u addr, Bit32u len, Bit8u *data);
-
-	bool loadControlROM(const char *filename);
-	bool loadPCMROM(const char *filename);
-	bool dumpTimbre(File *file, const TimbreParam *timbre, Bit32u addr);
-	int dumpTimbres(const char *filename, int start, int len);
-
-	bool initPCMList(Bit16u mapAddress, Bit16u count);
-	bool initRhythmTimbres(Bit16u mapAddress, Bit16u count);
-	bool initTimbres(Bit16u mapAddress, Bit16u offset, int startTimbre);
-	bool initRhythmTimbre(int drumNum, const Bit8u *mem, unsigned int memLen);
-	bool refreshSystem();
-
-protected:
-	int report(ReportType type, const void *reportData);
-	File *openFile(const char *filename, File::OpenMode mode);
-	void closeFile(File *file);
-	void printDebug(const char *fmt, ...) GCC_PRINTF(2, 3);
-
-public:
-	static Bit8u calcSysexChecksum(const Bit8u *data, Bit32u len, Bit8u checksum);
-
-	Synth();
-	~Synth();
-
-	// Used to initialise the MT-32. Must be called before any other function.
-	// Returns true if initialization was sucessful, otherwise returns false.
-	bool open(SynthProperties &useProp);
-
-	// Closes the MT-32 and deallocates any memory used by the synthesizer
-	void close(void);
-
-	// Sends a 4-byte MIDI message to the MT-32 for immediate playback
-	void playMsg(Bit32u msg);
-	void playMsgOnPart(unsigned char part, unsigned char code, unsigned char note, unsigned char velocity);
-
-	// Sends a string of Sysex commands to the MT-32 for immediate interpretation
-	// The length is in bytes
-	void playSysex(const Bit8u *sysex, Bit32u len);
-	void playSysexWithoutFraming(const Bit8u *sysex, Bit32u len);
-	void playSysexWithoutHeader(unsigned char device, unsigned char command, const Bit8u *sysex, Bit32u len);
-	void writeSysex(unsigned char channel, const Bit8u *sysex, Bit32u len);
-
-	// This callback routine is used to have the MT-32 generate samples to the specified
-	// output stream.  The length is in whole samples, not bytes. (I.E. in 16-bit stereo,
-	// one sample is 4 bytes)
-	void render(Bit16s * stream, Bit32u len);
-
-	const Partial *getPartial(unsigned int partialNum) const;
-
-	void readMemory(Bit32u addr, Bit32u len, Bit8u *data);
-
-	// partNum should be 0..7 for Part 1..8, or 8 for Rhythm
-	const Part *getPart(unsigned int partNum) const;
-};
-
-}
-
-#endif
diff --git a/sound/softsynth/mt32/tables.cpp b/sound/softsynth/mt32/tables.cpp
deleted file mode 100644
index eba4d2a520..0000000000
--- a/sound/softsynth/mt32/tables.cpp
+++ /dev/null
@@ -1,757 +0,0 @@
-/* Copyright (c) 2003-2005 Various contributors
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to
- * deal in the Software without restriction, including without limitation the
- * rights to use, copy, modify, merge, publish, distribute, sublicense, and/or
- * sell copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
- * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
- * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS
- * IN THE SOFTWARE.
- */
-
-#include <stdlib.h>
-#include <string.h>
-#include <math.h>
-
-#include "mt32emu.h"
-
-#if defined(MACOSX) || defined(SOLARIS) || defined(__MINGW32__)
-// Older versions of Mac OS X didn't supply a powf function, so using it
-// will cause a binary incompatibility when trying to run a binary built
-// on a newer OS X release on an olderr one. And Solaris 8 doesn't provide
-// powf, floorf, fabsf etc. at all.
-// Cross-compiled MinGW32 toolchains suffer from a cross-compile bug in
-// libstdc++. math/stubs.o should be empty, but it comes with a symbol for
-// powf, resulting in a linker error because of multiple definitions.
-// Hence we re-define them here. The only potential drawback is that it
-// might be a little bit slower this way.
-#define powf(x,y)	((float)pow(x,y))
-#define floorf(x)	((float)floor(x))
-#define fabsf(x)	((float)fabs(x))
-#endif
-
-#define FIXEDPOINT_MAKE(x, point) ((Bit32u)((1 << point) * x))
-
-namespace MT32Emu {
-
-//Amplitude time velocity follow exponential coefficients
-static const double tvcatconst[5] = {0.0, 0.002791309, 0.005942882, 0.012652792, 0.026938637};
-static const double tvcatmult[5] = {1.0, 1.072662811, 1.169129367, 1.288579123, 1.229630539};
-
-// These are division constants for the TVF depth key follow
-static const Bit32u depexp[5] = {3000, 950, 485, 255, 138};
-
-//Envelope time keyfollow exponential coefficients
-static const double tkcatconst[5] = {0.0, 0.005853144, 0.011148054, 0.019086143, 0.043333215};
-static const double tkcatmult[5] = {1.0, 1.058245688, 1.048488989, 1.016049301, 1.097538067};
-
-// Begin filter stuff
-
-// Pre-warp the coefficients of a numerator or denominator.
-// Note that a0 is assumed to be 1, so there is no wrapping
-// of it.
-static void prewarp(double *a1, double *a2, double fc, double fs) {
-	double wp;
-
-	wp = 2.0 * fs * tan(DOUBLE_PI * fc / fs);
-
-	*a2 = *a2 / (wp * wp);
-	*a1 = *a1 / wp;
-}
-
-// Transform the numerator and denominator coefficients
-// of s-domain biquad section into corresponding
-// z-domain coefficients.
-//
-//      Store the 4 IIR coefficients in array pointed by coef
-//      in following order:
-//             beta1, beta2    (denominator)
-//             alpha1, alpha2  (numerator)
-//
-// Arguments:
-//             a0-a2   - s-domain numerator coefficients
-//             b0-b2   - s-domain denominator coefficients
-//             k               - filter gain factor. initially set to 1
-//                                and modified by each biquad section in such
-//                                a way, as to make it the coefficient by
-//                                which to multiply the overall filter gain
-//                                in order to achieve a desired overall filter gain,
-//                                specified in initial value of k.
-//             fs             - sampling rate (Hz)
-//             coef    - array of z-domain coefficients to be filled in.
-//
-// Return:
-//             On return, set coef z-domain coefficients
-static void bilinear(double a0, double a1, double a2, double b0, double b1, double b2, double *k, double fs, float *coef) {
-	double ad, bd;
-
-	// alpha (Numerator in s-domain)
-	ad = 4. * a2 * fs * fs + 2. * a1 * fs + a0;
-	// beta (Denominator in s-domain)
-	bd = 4. * b2 * fs * fs + 2. * b1* fs + b0;
-
-	// update gain constant for this section
-	*k *= ad/bd;
-
-	// Denominator
-	*coef++ = (float)((2. * b0 - 8. * b2 * fs * fs) / bd);           // beta1
-	*coef++ = (float)((4. * b2 * fs * fs - 2. * b1 * fs + b0) / bd); // beta2
-
-	// Nominator
-	*coef++ = (float)((2. * a0 - 8. * a2 * fs * fs) / ad);           // alpha1
-	*coef = (float)((4. * a2 * fs * fs - 2. * a1 * fs + a0) / ad);   // alpha2
-}
-
-// a0-a2: numerator coefficients
-// b0-b2: denominator coefficients
-// fc: Filter cutoff frequency
-// fs: sampling rate
-// k: overall gain factor
-// coef: pointer to 4 iir coefficients
-static void szxform(double *a0, double *a1, double *a2, double *b0, double *b1, double *b2, double fc, double fs, double *k, float *coef) {
-	// Calculate a1 and a2 and overwrite the original values
-	prewarp(a1, a2, fc, fs);
-	prewarp(b1, b2, fc, fs);
-	bilinear(*a0, *a1, *a2, *b0, *b1, *b2, k, fs, coef);
-}
-
-static void initFilter(float fs, float fc, float *icoeff, float Q) {
-	float *coef;
-	double a0, a1, a2, b0, b1, b2;
-
-	double k = 1.5;    // Set overall filter gain factor
-	coef = icoeff + 1; // Skip k, or gain
-
-	// Section 1
-	a0 = 1.0;
-	a1 = 0;
-	a2 = 0;
-	b0 = 1.0;
-	b1 = 0.765367 / Q; // Divide by resonance or Q
-	b2 = 1.0;
-	szxform(&a0, &a1, &a2, &b0, &b1, &b2, fc, fs, &k, coef);
-	coef += 4;         // Point to next filter section
-
-	// Section 2
-	a0 = 1.0;
-	a1 = 0;
-	a2 = 0;
-	b0 = 1.0;
-	b1 = 1.847759 / Q;
-	b2 = 1.0;
-	szxform(&a0, &a1, &a2, &b0, &b1, &b2, fc, fs, &k, coef);
-
-	icoeff[0] = (float)k;
-}
-
-void Tables::initFiltCoeff(float samplerate) {
-	for (int j = 0; j < FILTERGRAN; j++) {
-		for (int res = 0; res < 31; res++) {
-			float tres = resonanceFactor[res];
-			initFilter((float)samplerate, (((float)(j+1.0)/FILTERGRAN)) * ((float)samplerate/2), filtCoeff[j][res], tres);
-		}
-	}
-}
-
-void Tables::initEnvelopes(float samplerate) {
-	for (int lf = 0; lf <= 100; lf++) {
-		float elf = (float)lf;
-
-		// General envelope
-		// This formula fits observation of the CM-32L by +/- 0.03s or so for the second time value in the filter,
-		// when all other times were 0 and all levels were 100. Note that variations occur depending on the level
-		// delta of the section, which we're not fully emulating.
-		float seconds = powf(2.0f, (elf / 8.0f) + 7.0f) / 32768.0f;
-		int samples = (int)(seconds * samplerate);
-		envTime[lf] = samples;
-
-		// Cap on envelope times depending on the level delta
-		if (elf == 0) {
-			envDeltaMaxTime[lf] = 63;
-		} else {
-			float cap = 11.0f * (float)log(elf) + 64;
-			if (cap > 100.0f) {
-				cap = 100.0f;
-			}
-			envDeltaMaxTime[lf] = (int)cap;
-		}
-
-
-		// This (approximately) represents the time durations when the target level is 0.
-		// Not sure why this is a special case, but it's seen to be from the real thing.
-		seconds = powf(2, (elf / 8.0f) + 6) / 32768.0f;
-		envDecayTime[lf]  = (int)(seconds * samplerate);
-
-		// I am certain of this:  Verified by hand LFO log
-		lfoPeriod[lf] = (Bit32u)(((float)samplerate) / (powf(1.088883372f, (float)lf) * 0.021236044f));
-	}
-}
-
-void Tables::initMT32ConstantTables(Synth *synth) {
-	int lf;
-	synth->printDebug("Initialising Pitch Tables");
-	for (lf = -108; lf <= 108; lf++) {
-		tvfKeyfollowMult[lf + 108] = (int)(256 * powf(2.0f, (float)(lf / 24.0f)));
-		//synth->printDebug("KT %d = %d", f, keytable[f+108]);
-	}
-
-	for (int res = 0; res < 31; res++) {
-		resonanceFactor[res] = powf((float)res / 30.0f, 5.0f) + 1.0f;
-	}
-
-	int period = 65536;
-
-	for (int ang = 0; ang < period; ang++) {
-		int halfang = (period / 2);
-		int angval = ang % halfang;
-		float tval = (((float)angval / (float)halfang) - 0.5f) * 2;
-		if (ang >= halfang)
-			tval = -tval;
-		sintable[ang] = (Bit16s)(tval * 50.0f) + 50;
-	}
-
-	int velt, dep;
-	float tempdep;
-	for (velt = 0; velt < 128; velt++) {
-		for (dep = 0; dep < 5; dep++) {
-			if (dep > 0) {
-				float ff = (float)(exp(3.5f * tvcatconst[dep] * (59.0f - (float)velt)) * tvcatmult[dep]);
-				tempdep = 256.0f * ff;
-				envTimeVelfollowMult[dep][velt] = (int)tempdep;
-				//if ((velt % 16) == 0) {
-				//	synth->printDebug("Key %d, depth %d, factor %d", velt, dep, (int)tempdep);
-				//}
-			} else
-				envTimeVelfollowMult[dep][velt] = 256;
-		}
-
-		for (dep = -7; dep < 8; dep++) {
-			float fldep = (float)abs(dep) / 7.0f;
-			fldep = powf(fldep,2.5f);
-			if (dep < 0)
-				fldep = fldep * -1.0f;
-			pwVelfollowAdd[dep+7][velt] = Bit32s((fldep * (float)velt * 100) / 128.0);
-		}
-	}
-
-	for (dep = 0; dep <= 100; dep++) {
-		for (velt = 0; velt < 128; velt++) {
-			float fdep = (float)dep * 0.000347013f; // Another MT-32 constant
-			float fv = ((float)velt - 64.0f)/7.26f;
-			float flogdep = powf(10, fdep * fv);
-			float fbase;
-
-			if (velt > 64)
-				synth->tables.tvfVelfollowMult[velt][dep] = (int)(flogdep * 256.0);
-			else {
-				//lff = 1 - (pow(((128.0 - (float)lf) / 64.0),.25) * ((float)velt / 96));
-				fbase = 1 - (powf(((float)dep / 100.0f),.25f) * ((float)(64-velt) / 96.0f));
-				synth->tables.tvfVelfollowMult[velt][dep] = (int)(fbase * 256.0);
-			}
-			//synth->printDebug("Filvel dep %d velt %d = %x", dep, velt, filveltable[velt][dep]);
-		}
-	}
-
-	for (lf = 0; lf < 128; lf++) {
-		float veloFract = lf / 127.0f;
-		for (int velsens = 0; velsens <= 100; velsens++) {
-			float sensFract = (velsens - 50) / 50.0f;
-			if (velsens < 50) {
-				tvaVelfollowMult[lf][velsens] = FIXEDPOINT_MAKE(1.0f / powf(2.0f, veloFract * -sensFract * 127.0f / 20.0f), 8);
-			} else {
-				tvaVelfollowMult[lf][velsens] = FIXEDPOINT_MAKE(1.0f / powf(2.0f, (1.0f - veloFract) * sensFract * 127.0f / 20.0f), 8);
-			}
-		}
-	}
-
-	for (lf = 0; lf <= 100; lf++) {
-		// Converts the 0-100 range used by the MT-32 to volume multiplier
-		volumeMult[lf] = FIXEDPOINT_MAKE(powf((float)lf / 100.0f, FLOAT_LN), 7);
-	}
-
-	for (lf = 0; lf <= 100; lf++) {
-		float mv = lf / 100.0f;
-		float pt = mv - 0.5f;
-		if (pt < 0)
-			pt = 0;
-
-		// Original (CC version)
-		//pwFactor[lf] = (int)(pt * 210.04f) + 128;
-
-		// Approximation from sample comparison
-		pwFactor[lf] = (int)(pt * 179.0f) + 128;
-	}
-
-	for (unsigned int i = 0; i < MAX_SAMPLE_OUTPUT; i++) {
-		int myRand;
-		myRand = rand();
-		//myRand = ((myRand - 16383) * 7168) >> 16;
-		// This one is slower but works with all values of RAND_MAX
-		myRand = (int)((myRand - RAND_MAX / 2) / (float)RAND_MAX * (7168 / 2));
-		//FIXME:KG: Original ultimately set the lowest two bits to 0, for no obvious reason
-		noiseBuf[i] = (Bit16s)myRand;
-	}
-
-	float tdist;
-	float padjtable[51];
-	for (lf = 0; lf <= 50; lf++) {
-		if (lf == 0)
-			padjtable[lf] = 7;
-		else if (lf == 1)
-			padjtable[lf] = 6;
-		else if (lf == 2)
-			padjtable[lf] = 5;
-		else if (lf == 3)
-			padjtable[lf] = 4;
-		else if (lf == 4)
-			padjtable[lf] = 4 - (0.333333f);
-		else if (lf == 5)
-			padjtable[lf] = 4 - (0.333333f * 2);
-		else if (lf == 6)
-			padjtable[lf] = 3;
-		else if ((lf > 6) && (lf <= 12)) {
-			tdist = (lf-6.0f) / 6.0f;
-			padjtable[lf] = 3.0f - tdist;
-		} else if ((lf > 12) && (lf <= 25)) {
-			tdist = (lf - 12.0f) / 13.0f;
-			padjtable[lf] = 2.0f - tdist;
-		} else {
-			tdist = (lf - 25.0f) / 25.0f;
-			padjtable[lf] = 1.0f - tdist;
-		}
-		//synth->printDebug("lf %d = padj %f", lf, (double)padjtable[lf]);
-	}
-
-	float lfp, depf, finalval, tlf;
-	int depat, pval, depti;
-	for (lf = 0; lf <= 10; lf++) {
-		// I believe the depth is cubed or something
-
-		for (depat = 0; depat <= 100; depat++) {
-			if (lf > 0) {
-				depti = abs(depat - 50);
-				tlf = (float)lf - padjtable[depti];
-				if (tlf < 0)
-					tlf = 0;
-				lfp = (float)exp(0.713619942f * tlf) / 407.4945111f;
-
-				if (depat < 50)
-					finalval = 4096.0f * powf(2, -lfp);
-				else
-					finalval = 4096.0f * powf(2, lfp);
-				pval = (int)finalval;
-
-				pitchEnvVal[lf][depat] = pval;
-				//synth->printDebug("lf %d depat %d pval %d tlf %f lfp %f", lf,depat,pval, (double)tlf, (double)lfp);
-			} else {
-				pitchEnvVal[lf][depat] = 4096;
-				//synth->printDebug("lf %d depat %d pval 4096", lf, depat);
-			}
-		}
-	}
-	for (lf = 0; lf <= 100; lf++) {
-		// It's linear - verified on MT-32 - one of the few things linear
-		lfp = ((float)lf * 0.1904f) / 310.55f;
-
-		for (depat = 0; depat <= 100; depat++) {
-			depf = ((float)depat - 50.0f) / 50.0f;
-			//finalval = pow(2, lfp * depf * .5);
-			finalval = 4096.0f + (4096.0f * lfp * depf);
-
-			pval = (int)finalval;
-
-			lfoShift[lf][depat] = pval;
-
-			//synth->printDebug("lf %d depat %d pval %x", lf,depat,pval);
-		}
-	}
-
-	for (lf = 0; lf <= 12; lf++) {
-		for (int distval = 0; distval < 128; distval++) {
-			float amplog, dval;
-			if (lf == 0) {
-				amplog = 0;
-				dval = 1;
-				tvaBiasMult[lf][distval] = 256;
-			} else {
-				/*
-				amplog = powf(1.431817011f, (float)lf) / FLOAT_PI;
-				dval = ((128.0f - (float)distval) / 128.0f);
-				amplog = exp(amplog);
-				dval = powf(amplog, dval) / amplog;
-				tvaBiasMult[lf][distval] = (int)(dval * 256.0);
-				*/
-				// Lets assume for a second it's linear
-
-				// Distance of full volume reduction
-				amplog = (float)(12.0f / (float)lf) * 24.0f;
-				if (distval > amplog) {
-					tvaBiasMult[lf][distval] = 0;
-				} else {
-					dval = (amplog - (float)distval) / amplog;
-					tvaBiasMult[lf][distval] = (int)(dval * 256.0f);
-				}
-			}
-			//synth->printDebug("Ampbias lf %d distval %d = %f (%x) %f", lf, distval, (double)dval, tvaBiasMult[lf][distval],(double)amplog);
-		}
-	}
-
-	for (lf = 0; lf <= 14; lf++) {
-		for (int distval = 0; distval < 128; distval++) {
-			float filval = fabsf((float)((lf - 7) * 12) / 7.0f);
-			float amplog, dval;
-			if (lf == 7) {
-				amplog = 0;
-				dval = 1;
-				tvfBiasMult[lf][distval] = 256;
-			} else {
-				//amplog = pow(1.431817011, filval) / FLOAT_PI;
-				amplog = powf(1.531817011f, filval) / FLOAT_PI;
-				dval = (128.0f - (float)distval) / 128.0f;
-				amplog = (float)exp(amplog);
-				dval = powf(amplog,dval)/amplog;
-				if (lf < 8) {
-					tvfBiasMult[lf][distval] = (int)(dval * 256.0f);
-				} else {
-					dval = powf(dval, 0.3333333f);
-					if (dval < 0.01f)
-						dval = 0.01f;
-					dval = 1 / dval;
-					tvfBiasMult[lf][distval] = (int)(dval * 256.0f);
-				}
-			}
-			//synth->printDebug("Fbias lf %d distval %d = %f (%x) %f", lf, distval, (double)dval, tvfBiasMult[lf][distval],(double)amplog);
-		}
-	}
-}
-
-// Per-note table initialisation follows
-
-static void initSaw(NoteLookup *noteLookup, Bit32s div2) {
-	int tmpdiv = div2 << 16;
-	for (int rsaw = 0; rsaw <= 100; rsaw++) {
-		float fsaw;
-		if (rsaw < 50)
-			fsaw = 50.0f;
-		else
-			fsaw = (float)rsaw;
-
-		//(66 - (((A8 - 50) / 50) ^ 0.63) * 50) / 132
-		float sawfact = (66.0f - (powf((fsaw - 50.0f) / 50.0f, 0.63f) * 50.0f)) / 132.0f;
-		noteLookup->sawTable[rsaw] = (int)(sawfact * (float)tmpdiv) >> 16;
-		//synth->printDebug("F %d divtable %d saw %d sawtable %d", f, div, rsaw, sawtable[f][rsaw]);
-	}
-}
-
-static void initDep(KeyLookup *keyLookup, float f) {
-	for (int dep = 0; dep < 5; dep++) {
-		if (dep == 0) {
-			keyLookup->envDepthMult[dep] = 256;
-			keyLookup->envTimeMult[dep] = 256;
-		} else {
-			float depfac = 3000.0f;
-			float ff, tempdep;
-			depfac = (float)depexp[dep];
-
-			ff = (f - (float)MIDDLEC) / depfac;
-			tempdep = powf(2, ff) * 256.0f;
-			keyLookup->envDepthMult[dep] = (int)tempdep;
-
-			ff = (float)(exp(tkcatconst[dep] * ((float)MIDDLEC - f)) * tkcatmult[dep]);
-			keyLookup->envTimeMult[dep] = (int)(ff * 256.0f);
-		}
-	}
-	//synth->printDebug("F %f d1 %x d2 %x d3 %x d4 %x d5 %x", (double)f, noteLookup->fildepTable[0], noteLookup->fildepTable[1], noteLookup->fildepTable[2], noteLookup->fildepTable[3], noteLookup->fildepTable[4]);
-}
-
-Bit16s Tables::clampWF(Synth *synth, const char *n, float ampVal, double input) {
-	Bit32s x = (Bit32s)(input * ampVal);
-	if (x < -ampVal - 1) {
-		synth->printDebug("%s==%d<-WGAMP-1!", n, x);
-		x = (Bit32s)(-ampVal - 1);
-	} else if (x > ampVal) {
-		synth->printDebug("%s==%d>WGAMP!", n, x);
-		x = (Bit32s)ampVal;
-	}
-	return (Bit16s)x;
-}
-
-File *Tables::initWave(Synth *synth, NoteLookup *noteLookup, float ampVal, float div2, File *file) {
-	int iDiv2 = (int)div2;
-	noteLookup->waveformSize[0] = iDiv2 << 1;
-	noteLookup->waveformSize[1] = iDiv2 << 1;
-	noteLookup->waveformSize[2] = iDiv2 << 2;
-	for (int i = 0; i < 3; i++) {
-		if (noteLookup->waveforms[i] == NULL) {
-			noteLookup->waveforms[i] = new Bit16s[noteLookup->waveformSize[i]];
-		}
-	}
-	if (file != NULL) {
-		for (int i = 0; i < 3 && file != NULL; i++) {
-			size_t len = noteLookup->waveformSize[i];
-			for (unsigned int j = 0; j < len; j++) {
-				if (!file->readBit16u((Bit16u *)&noteLookup->waveforms[i][j])) {
-					synth->printDebug("Error reading wave file cache!");
-					file->close();
-					file = NULL;
-					break;
-				}
-			}
-		}
-	}
-	if (file == NULL) {
-		double sd = DOUBLE_PI / div2;
-
-		for (int fa = 0; fa < (iDiv2 << 1); fa++) {
-			// sa ranges from 0 to 2PI
-			double sa = fa * sd;
-
-			// Calculate a sample for the bandlimited sawtooth wave
-			double saw = 0.0;
-			int sincs = iDiv2 >> 1;
-			double sinus = 1.0;
-			for (int sincNum = 1; sincNum <= sincs; sincNum++) {
-				saw += sin(sinus * sa) / sinus;
-				sinus++;
-			}
-
-			// This works pretty well
-			// Multiplied by 0.84 so that the spikes caused by bandlimiting don't overdrive the amplitude
-			noteLookup->waveforms[0][fa] = clampWF(synth, "saw", ampVal, -saw / (0.5 * DOUBLE_PI) * 0.84);
-			noteLookup->waveforms[1][fa] = clampWF(synth, "cos", ampVal, -cos(sa / 2.0));
-			noteLookup->waveforms[2][fa * 2] = clampWF(synth, "cosoff_0", ampVal, -cos(sa - DOUBLE_PI));
-			noteLookup->waveforms[2][fa * 2 + 1] = clampWF(synth, "cosoff_1", ampVal, -cos((sa + (sd / 2)) - DOUBLE_PI));
-		}
-	}
-	return file;
-}
-
-static void initFiltTable(NoteLookup *noteLookup, float freq, float rate) {
-	for (int tr = 0; tr <= 200; tr++) {
-		float ftr = (float)tr;
-
-		// Verified exact on MT-32
-		if (tr > 100)
-			ftr = 100.0f + (powf((ftr - 100.0f) / 100.0f, 3.0f) * 100.0f);
-
-		// I think this is the one
-		float brsq = powf(10.0f, (tr / 50.0f) - 1.0f);
-		noteLookup->filtTable[0][tr] = (int)((freq * brsq) / (rate / 2) * FILTERGRAN);
-		if (noteLookup->filtTable[0][tr]>=((FILTERGRAN*15)/16))
-			noteLookup->filtTable[0][tr] = ((FILTERGRAN*15)/16);
-
-		float brsa = powf(10.0f, ((tr / 55.0f) - 1.0f)) / 2.0f;
-		noteLookup->filtTable[1][tr] = (int)((freq * brsa) / (rate / 2) * FILTERGRAN);
-		if (noteLookup->filtTable[1][tr]>=((FILTERGRAN*15)/16))
-			noteLookup->filtTable[1][tr] = ((FILTERGRAN*15)/16);
-	}
-}
-
-static void initNFiltTable(NoteLookup *noteLookup, float freq, float rate) {
-	for (int cf = 0; cf <= 100; cf++) {
-		float cfmult = (float)cf;
-
-		for (int tf = 0;tf <= 100; tf++) {
-			float tfadd = (float)tf;
-
-			//float freqsum = exp((cfmult + tfadd) / 30.0f) / 4.0f;
-			//float freqsum = 0.15f * exp(0.45f * ((cfmult + tfadd) / 10.0f));
-
-			float freqsum = powf(2.0f, ((cfmult + tfadd) - 40.0f) / 16.0f);
-
-			noteLookup->nfiltTable[cf][tf] = (int)((freq * freqsum) / (rate / 2) * FILTERGRAN);
-			if (noteLookup->nfiltTable[cf][tf] >= ((FILTERGRAN * 15) / 16))
-				noteLookup->nfiltTable[cf][tf] = ((FILTERGRAN * 15) / 16);
-		}
-	}
-}
-
-File *Tables::initNote(Synth *synth, NoteLookup *noteLookup, float note, float rate, float masterTune, PCMWaveEntry *pcmWaves, File *file) {
-	float freq = (float)(masterTune * pow(2.0, ((double)note - MIDDLEA) / 12.0));
-	float div2 = rate * 2.0f / freq;
-	noteLookup->div2 = (int)div2;
-
-	if (noteLookup->div2 == 0)
-		noteLookup->div2 = 1;
-
-	initSaw(noteLookup, noteLookup->div2);
-
-	//synth->printDebug("Note %f; freq=%f, div=%f", (double)note, (double)freq, (double) rate / freq);
-	file = initWave(synth, noteLookup, WGAMP, div2, file);
-
-	// Create the pitch tables
-	if (noteLookup->wavTable == NULL)
-		noteLookup->wavTable = new Bit32u[synth->controlROMMap->pcmCount];
-	double rateMult = 32000.0 / rate;
-	double tuner = freq * 65536.0f;
-	for (int pc = 0; pc < synth->controlROMMap->pcmCount; pc++) {
-		noteLookup->wavTable[pc] = (int)(tuner / pcmWaves[pc].tune * rateMult);
-	}
-
-	initFiltTable(noteLookup, freq, rate);
-	initNFiltTable(noteLookup, freq, rate);
-	return file;
-}
-
-bool Tables::initNotes(Synth *synth, PCMWaveEntry *pcmWaves, float rate, float masterTune) {
-	const char *NoteNames[12] = {
-		"C ", "C#", "D ", "D#", "E ", "F ", "F#", "G ", "G#", "A ", "A#", "B "
-	};
-	char filename[64];
-	int intRate = (int)rate;
-	char version[4] = {0, 0, 0, 5};
-	sprintf(filename, "waveformcache-%d-%.2f.raw", intRate, (double)masterTune);
-
-	File *file = NULL;
-	char header[20];
-	memcpy(header, "MT32WAVE", 8);
-	int pos = 8;
-	// Version...
-	for (int i = 0; i < 4; i++)
-		header[pos++] = version[i];
-	header[pos++] = (char)((intRate >> 24) & 0xFF);
-	header[pos++] = (char)((intRate >> 16) & 0xFF);
-	header[pos++] = (char)((intRate >> 8) & 0xFF);
-	header[pos++] = (char)(intRate & 0xFF);
-	int intTuning = (int)masterTune;
-	header[pos++] = (char)((intTuning >> 8) & 0xFF);
-	header[pos++] = (char)(intTuning & 0xFF);
-	header[pos++] = 0;
-	header[pos] = (char)((masterTune - intTuning) * 10);
-#if MT32EMU_WAVECACHEMODE < 2
-	bool reading = false;
-	file = synth->openFile(filename, File::OpenMode_read);
-	if (file != NULL) {
-		char fileHeader[20];
-		if (file->read(fileHeader, 20) == 20) {
-			if (memcmp(fileHeader, header, 20) == 0) {
-				Bit16u endianCheck;
-				if (file->readBit16u(&endianCheck)) {
-					if (endianCheck == 1) {
-						reading = true;
-					} else {
-						synth->printDebug("Endian check in %s does not match expected", filename);
-					}
-				} else {
-					synth->printDebug("Unable to read endian check in %s", filename);
-				}
-			} else {
-				synth->printDebug("Header of %s does not match expected", filename);
-			}
-		} else {
-			synth->printDebug("Error reading 16 bytes of %s", filename);
-		}
-		if (!reading) {
-			file->close();
-			file = NULL;
-		}
-	} else {
-		synth->printDebug("Unable to open %s for reading", filename);
-	}
-#endif
-
-	float progress = 0.0f;
-	bool abort = false;
-	synth->report(ReportType_progressInit, &progress);
-	for (int f = LOWEST_NOTE; f <= HIGHEST_NOTE; f++) {
-		synth->printDebug("Initialising note %s%d", NoteNames[f % 12], (f / 12) - 2);
-		NoteLookup *noteLookup = &noteLookups[f - LOWEST_NOTE];
-		file = initNote(synth, noteLookup, (float)f, rate, masterTune, pcmWaves, file);
-		progress = (f - LOWEST_NOTE + 1) / (float)NUM_NOTES;
-		abort = synth->report(ReportType_progressInit, &progress) != 0;
-		if (abort)
-			break;
-	}
-
-#if MT32EMU_WAVECACHEMODE == 0 || MT32EMU_WAVECACHEMODE == 2
-	if (file == NULL) {
-		file = synth->openFile(filename, File::OpenMode_write);
-		if (file != NULL) {
-			if (file->write(header, 20) == 20 && file->writeBit16u(1)) {
-				for (int f = 0; f < NUM_NOTES; f++) {
-					for (int i = 0; i < 3 && file != NULL; i++) {
-						int len = noteLookups[f].waveformSize[i];
-						for (int j = 0; j < len; j++) {
-							if (!file->writeBit16u(noteLookups[f].waveforms[i][j])) {
-								synth->printDebug("Error writing waveform cache file");
-								file->close();
-								file = NULL;
-								break;
-							}
-						}
-					}
-				}
-			} else {
-				synth->printDebug("Error writing 16-byte header to %s - won't continue saving", filename);
-			}
-		} else {
-			synth->printDebug("Unable to open %s for writing - won't be created", filename);
-		}
-	}
-#endif
-
-	if (file != NULL)
-		synth->closeFile(file);
-	return !abort;
-}
-
-void Tables::freeNotes() {
-	for (int t = 0; t < 3; t++) {
-		for (int m = 0; m < NUM_NOTES; m++) {
-			if (noteLookups[m].waveforms[t] != NULL) {
-				delete[] noteLookups[m].waveforms[t];
-				noteLookups[m].waveforms[t] = NULL;
-				noteLookups[m].waveformSize[t] = 0;
-			}
-			if (noteLookups[m].wavTable != NULL) {
-				delete[] noteLookups[m].wavTable;
-				noteLookups[m].wavTable = NULL;
-			}
-		}
-	}
-	initialisedMasterTune = 0.0f;
-}
-
-Tables::Tables() {
-	initialisedSampleRate = 0.0f;
-	initialisedMasterTune = 0.0f;
-	memset(&noteLookups, 0, sizeof(noteLookups));
-}
-
-bool Tables::init(Synth *synth, PCMWaveEntry *pcmWaves, float sampleRate, float masterTune) {
-	if (sampleRate <= 0.0f) {
-		synth->printDebug("Bad sampleRate (%f <= 0.0f)", (double)sampleRate);
-		return false;
-	}
-	if (initialisedSampleRate == 0.0f) {
-		initMT32ConstantTables(synth);
-	}
-	if (initialisedSampleRate != sampleRate) {
-		initFiltCoeff(sampleRate);
-		initEnvelopes(sampleRate);
-		for (int key = 12; key <= 108; key++) {
-			initDep(&keyLookups[key - 12], (float)key);
-		}
-	}
-	if (initialisedSampleRate != sampleRate || initialisedMasterTune != masterTune) {
-		freeNotes();
-		if (!initNotes(synth, pcmWaves, sampleRate, masterTune)) {
-			return false;
-		}
-		initialisedSampleRate = sampleRate;
-		initialisedMasterTune = masterTune;
-	}
-	return true;
-}
-
-}
diff --git a/sound/softsynth/mt32/tables.h b/sound/softsynth/mt32/tables.h
deleted file mode 100644
index d9af5114b2..0000000000
--- a/sound/softsynth/mt32/tables.h
+++ /dev/null
@@ -1,116 +0,0 @@
-/* Copyright (c) 2003-2005 Various contributors
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to
- * deal in the Software without restriction, including without limitation the
- * rights to use, copy, modify, merge, publish, distribute, sublicense, and/or
- * sell copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
- * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
- * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS
- * IN THE SOFTWARE.
- */
-
-#ifndef MT32EMU_TABLES_H
-#define MT32EMU_TABLES_H
-
-namespace MT32Emu {
-
-// Mathematical constants
-const double DOUBLE_PI = 3.1415926535897932384626433832795;
-const double DOUBLE_LN = 2.3025850929940456840179914546844;
-const float FLOAT_PI = 3.1415926535897932384626433832795f;
-const float FLOAT_LN = 2.3025850929940456840179914546844f;
-
-// Filter settings
-const int FILTERGRAN = 512;
-
-// Amplitude of waveform generator
-// FIXME: This value is the amplitude possible whilst avoiding
-// overdriven values immediately after filtering when playing
-// back SQ3MT.MID. Needs to be checked.
-const int WGAMP = 12382;
-
-const int MIDDLEC = 60;
-const int MIDDLEA = 69; // By this I mean "A above middle C"
-
-// FIXME:KG: may only need to do 12 to 108
-// 12..108 is the range allowed by note on commands, but the key can be modified by pitch keyfollow
-// and adjustment for timbre pitch, so the results can be outside that range.
-// Should we move it (by octave) into the 12..108 range, or keep it in 0..127 range,
-// or something else altogether?
-const int LOWEST_NOTE = 12;
-const int HIGHEST_NOTE = 127;
-const int NUM_NOTES = HIGHEST_NOTE - LOWEST_NOTE + 1; // Number of slots for note LUT
-
-class Synth;
-
-struct NoteLookup {
-	Bit32u div2;
-	Bit32u *wavTable;
-	Bit32s sawTable[101];
-	int filtTable[2][201];
-	int nfiltTable[101][101];
-	Bit16s *waveforms[3];
-	Bit32u waveformSize[3];
-};
-
-struct KeyLookup {
-	Bit32s envTimeMult[5]; // For envelope time adjustment for key pressed
-	Bit32s envDepthMult[5];
-};
-
-class Tables {
-	float initialisedSampleRate;
-	float initialisedMasterTune;
-	void initMT32ConstantTables(Synth *synth);
-	static Bit16s clampWF(Synth *synth, const char *n, float ampVal, double input);
-	static File *initWave(Synth *synth, NoteLookup *noteLookup, float ampsize, float div2, File *file);
-	bool initNotes(Synth *synth, PCMWaveEntry *pcmWaves, float rate, float tuning);
-	void initEnvelopes(float sampleRate);
-	void initFiltCoeff(float samplerate);
-public:
-	// Constant LUTs
-	Bit32s tvfKeyfollowMult[217];
-	Bit32s tvfVelfollowMult[128][101];
-	Bit32s tvfBiasMult[15][128];
-	Bit32u tvaVelfollowMult[128][101];
-	Bit32s tvaBiasMult[13][128];
-	Bit16s noiseBuf[MAX_SAMPLE_OUTPUT];
-	Bit16s sintable[65536];
-	Bit32s pitchEnvVal[16][101];
-	Bit32s envTimeVelfollowMult[5][128];
-	Bit32s pwVelfollowAdd[15][128];
-	float resonanceFactor[31];
-	Bit32u lfoShift[101][101];
-	Bit32s pwFactor[101];
-	Bit32s volumeMult[101];
-
-	// LUTs varying with sample rate
-	Bit32u envTime[101];
-	Bit32u envDeltaMaxTime[101];
-	Bit32u envDecayTime[101];
-	Bit32u lfoPeriod[101];
-	float filtCoeff[FILTERGRAN][31][8];
-
-	// Various LUTs for each note and key
-	NoteLookup noteLookups[NUM_NOTES];
-	KeyLookup keyLookups[97];
-
-	Tables();
-	bool init(Synth *synth, PCMWaveEntry *pcmWaves, float sampleRate, float masterTune);
-	File *initNote(Synth *synth, NoteLookup *noteLookup, float note, float rate, float tuning, PCMWaveEntry *pcmWaves, File *file);
-	void freeNotes();
-};
-
-}
-
-#endif
diff --git a/sound/softsynth/opl/dbopl.cpp b/sound/softsynth/opl/dbopl.cpp
deleted file mode 100644
index 47e263b6b9..0000000000
--- a/sound/softsynth/opl/dbopl.cpp
+++ /dev/null
@@ -1,1536 +0,0 @@
-/*
- *  Copyright (C) 2002-2010  The DOSBox Team
- *
- *  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., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
- */
-
-/*
-	DOSBox implementation of a combined Yamaha YMF262 and Yamaha YM3812 emulator.
-	Enabling the opl3 bit will switch the emulator to stereo opl3 output instead of regular mono opl2
-	Except for the table generation it's all integer math
-	Can choose different types of generators, using muls and bigger tables, try different ones for slower platforms
-	The generation was based on the MAME implementation but tried to have it use less memory and be faster in general
-	MAME uses much bigger envelope tables and this will be the biggest cause of it sounding different at times
-
-	//TODO Don't delay first operator 1 sample in opl3 mode
-	//TODO Maybe not use class method pointers but a regular function pointers with operator as first parameter
-	//TODO Fix panning for the Percussion channels, would any opl3 player use it and actually really change it though?
-	//TODO Check if having the same accuracy in all frequency multipliers sounds better or not
-
-	//DUNNO Keyon in 4op, switch to 2op without keyoff.
-*/
-
-// Last synch with DOSBox SVN trunk r3556
-
-#include "dbopl.h"
-
-#ifndef DISABLE_DOSBOX_OPL
-
-namespace OPL {
-namespace DOSBox {
-
-#ifndef PI
-#define PI 3.14159265358979323846
-#endif
-
-namespace DBOPL {
-
-#define OPLRATE		((double)(14318180.0 / 288.0))
-#define TREMOLO_TABLE 52
-
-//Try to use most precision for frequencies
-//Else try to keep different waves in synch
-//#define WAVE_PRECISION	1
-#ifndef WAVE_PRECISION
-//Wave bits available in the top of the 32bit range
-//Original adlib uses 10.10, we use 10.22
-#define WAVE_BITS	10
-#else
-//Need some extra bits at the top to have room for octaves and frequency multiplier
-//We support to 8 times lower rate
-//128 * 15 * 8 = 15350, 2^13.9, so need 14 bits
-#define WAVE_BITS	14
-#endif
-#define WAVE_SH		( 32 - WAVE_BITS )
-#define WAVE_MASK	( ( 1 << WAVE_SH ) - 1 )
-
-//Use the same accuracy as the waves
-#define LFO_SH ( WAVE_SH - 10 )
-//LFO is controlled by our tremolo 256 sample limit
-#define LFO_MAX ( 256 << ( LFO_SH ) )
-
-
-//Maximum amount of attenuation bits
-//Envelope goes to 511, 9 bits
-#if (DBOPL_WAVE == WAVE_TABLEMUL )
-//Uses the value directly
-#define ENV_BITS	( 9 )
-#else
-//Add 3 bits here for more accuracy and would have to be shifted up either way
-#define ENV_BITS	( 9 )
-#endif
-//Limits of the envelope with those bits and when the envelope goes silent
-#define ENV_MIN		0
-#define ENV_EXTRA	( ENV_BITS - 9 )
-#define ENV_MAX		( 511 << ENV_EXTRA )
-#define ENV_LIMIT	( ( 12 * 256) >> ( 3 - ENV_EXTRA ) )
-#define ENV_SILENT( _X_ ) ( (_X_) >= ENV_LIMIT )
-
-//Attack/decay/release rate counter shift
-#define RATE_SH		24
-#define RATE_MASK	( ( 1 << RATE_SH ) - 1 )
-//Has to fit within 16bit lookuptable
-#define MUL_SH		16
-
-//Check some ranges
-#if ENV_EXTRA > 3
-#error Too many envelope bits
-#endif
-
-
-//How much to substract from the base value for the final attenuation
-static const Bit8u KslCreateTable[16] = {
-	//0 will always be be lower than 7 * 8
-	64, 32, 24, 19,
-	16, 12, 11, 10,
-	 8,  6,  5,  4,
-	 3,  2,  1,  0,
-};
-
-#define M(_X_) ((Bit8u)( (_X_) * 2))
-static const Bit8u FreqCreateTable[16] = {
-	M(0.5), M(1 ), M(2 ), M(3 ), M(4 ), M(5 ), M(6 ), M(7 ),
-	M(8  ), M(9 ), M(10), M(10), M(12), M(12), M(15), M(15)
-};
-#undef M
-
-//We're not including the highest attack rate, that gets a special value
-static const Bit8u AttackSamplesTable[13] = {
-	69, 55, 46, 40,
-	35, 29, 23, 20,
-	19, 15, 11, 10,
-	9
-};
-//On a real opl these values take 8 samples to reach and are based upon larger tables
-static const Bit8u EnvelopeIncreaseTable[13] = {
-	4,  5,  6,  7,
-	8, 10, 12, 14,
-	16, 20, 24, 28,
-	32,
-};
-
-#if ( DBOPL_WAVE == WAVE_HANDLER ) || ( DBOPL_WAVE == WAVE_TABLELOG )
-static Bit16u ExpTable[ 256 ];
-#endif
-
-#if ( DBOPL_WAVE == WAVE_HANDLER )
-//PI table used by WAVEHANDLER
-static Bit16u SinTable[ 512 ];
-#endif
-
-#if ( DBOPL_WAVE > WAVE_HANDLER )
-//Layout of the waveform table in 512 entry intervals
-//With overlapping waves we reduce the table to half it's size
-
-//	|    |//\\|____|WAV7|//__|/\  |____|/\/\|
-//	|\\//|    |    |WAV7|    |  \/|    |    |
-//	|06  |0126|17  |7   |3   |4   |4 5 |5   |
-
-//6 is just 0 shifted and masked
-
-static Bit16s WaveTable[ 8 * 512 ];
-//Distance into WaveTable the wave starts
-static const Bit16u WaveBaseTable[8] = {
-	0x000, 0x200, 0x200, 0x800,
-	0xa00, 0xc00, 0x100, 0x400,
-
-};
-//Mask the counter with this
-static const Bit16u WaveMaskTable[8] = {
-	1023, 1023, 511, 511,
-	1023, 1023, 512, 1023,
-};
-
-//Where to start the counter on at keyon
-static const Bit16u WaveStartTable[8] = {
-	512, 0, 0, 0,
-	0, 512, 512, 256,
-};
-#endif
-
-#if ( DBOPL_WAVE == WAVE_TABLEMUL )
-static Bit16u MulTable[ 384 ];
-#endif
-
-static Bit8u KslTable[ 8 * 16 ];
-static Bit8u TremoloTable[ TREMOLO_TABLE ];
-//Start of a channel behind the chip struct start
-static Bit16u ChanOffsetTable[32];
-//Start of an operator behind the chip struct start
-static Bit16u OpOffsetTable[64];
-
-//The lower bits are the shift of the operator vibrato value
-//The highest bit is right shifted to generate -1 or 0 for negation
-//So taking the highest input value of 7 this gives 3, 7, 3, 0, -3, -7, -3, 0
-static const Bit8s VibratoTable[ 8 ] = {
-	1 - 0x00, 0 - 0x00, 1 - 0x00, 30 - 0x00,
-	1 - 0x80, 0 - 0x80, 1 - 0x80, 30 - 0x80
-};
-
-//Shift strength for the ksl value determined by ksl strength
-static const Bit8u KslShiftTable[4] = {
-	31,1,2,0
-};
-
-//Generate a table index and table shift value using input value from a selected rate
-static void EnvelopeSelect( Bit8u val, Bit8u& index, Bit8u& shift ) {
-	if ( val < 13 * 4 ) {				//Rate 0 - 12
-		shift = 12 - ( val >> 2 );
-		index = val & 3;
-	} else if ( val < 15 * 4 ) {		//rate 13 - 14
-		shift = 0;
-		index = val - 12 * 4;
-	} else {							//rate 15 and up
-		shift = 0;
-		index = 12;
-	}
-}
-
-#if ( DBOPL_WAVE == WAVE_HANDLER )
-/*
-	Generate the different waveforms out of the sine/exponetial table using handlers
-*/
-static inline Bits MakeVolume( Bitu wave, Bitu volume ) {
-	Bitu total = wave + volume;
-	Bitu index = total & 0xff;
-	Bitu sig = ExpTable[ index ];
-	Bitu exp = total >> 8;
-#if 0
-	//Check if we overflow the 31 shift limit
-	if ( exp >= 32 ) {
-		LOG_MSG( "WTF %d %d", total, exp );
-	}
-#endif
-	return (sig >> exp);
-}
-
-static Bits DB_FASTCALL WaveForm0( Bitu i, Bitu volume ) {
-	Bits neg = 0 - (( i >> 9) & 1);//Create ~0 or 0
-	Bitu wave = SinTable[i & 511];
-	return (MakeVolume( wave, volume ) ^ neg) - neg;
-}
-static Bits DB_FASTCALL WaveForm1( Bitu i, Bitu volume ) {
-	Bit32u wave = SinTable[i & 511];
-	wave |= ( ( (i ^ 512 ) & 512) - 1) >> ( 32 - 12 );
-	return MakeVolume( wave, volume );
-}
-static Bits DB_FASTCALL WaveForm2( Bitu i, Bitu volume ) {
-	Bitu wave = SinTable[i & 511];
-	return MakeVolume( wave, volume );
-}
-static Bits DB_FASTCALL WaveForm3( Bitu i, Bitu volume ) {
-	Bitu wave = SinTable[i & 255];
-	wave |= ( ( (i ^ 256 ) & 256) - 1) >> ( 32 - 12 );
-	return MakeVolume( wave, volume );
-}
-static Bits DB_FASTCALL WaveForm4( Bitu i, Bitu volume ) {
-	//Twice as fast
-	i <<= 1;
-	Bits neg = 0 - (( i >> 9) & 1);//Create ~0 or 0
-	Bitu wave = SinTable[i & 511];
-	wave |= ( ( (i ^ 512 ) & 512) - 1) >> ( 32 - 12 );
-	return (MakeVolume( wave, volume ) ^ neg) - neg;
-}
-static Bits DB_FASTCALL WaveForm5( Bitu i, Bitu volume ) {
-	//Twice as fast
-	i <<= 1;
-	Bitu wave = SinTable[i & 511];
-	wave |= ( ( (i ^ 512 ) & 512) - 1) >> ( 32 - 12 );
-	return MakeVolume( wave, volume );
-}
-static Bits DB_FASTCALL WaveForm6( Bitu i, Bitu volume ) {
-	Bits neg = 0 - (( i >> 9) & 1);//Create ~0 or 0
-	return (MakeVolume( 0, volume ) ^ neg) - neg;
-}
-static Bits DB_FASTCALL WaveForm7( Bitu i, Bitu volume ) {
-	//Negative is reversed here
-	Bits neg = (( i >> 9) & 1) - 1;
-	Bitu wave = (i << 3);
-	//When negative the volume also runs backwards
-	wave = ((wave ^ neg) - neg) & 4095;
-	return (MakeVolume( wave, volume ) ^ neg) - neg;
-}
-
-static const WaveHandler WaveHandlerTable[8] = {
-	WaveForm0, WaveForm1, WaveForm2, WaveForm3,
-	WaveForm4, WaveForm5, WaveForm6, WaveForm7
-};
-
-#endif
-
-/*
-	Operator
-*/
-
-//We zero out when rate == 0
-inline void Operator::UpdateAttack( const Chip* chip ) {
-	Bit8u rate = reg60 >> 4;
-	if ( rate ) {
-		Bit8u val = (rate << 2) + ksr;
-		attackAdd = chip->attackRates[ val ];
-		rateZero &= ~(1 << ATTACK);
-	} else {
-		attackAdd = 0;
-		rateZero |= (1 << ATTACK);
-	}
-}
-inline void Operator::UpdateDecay( const Chip* chip ) {
-	Bit8u rate = reg60 & 0xf;
-	if ( rate ) {
-		Bit8u val = (rate << 2) + ksr;
-		decayAdd = chip->linearRates[ val ];
-		rateZero &= ~(1 << DECAY);
-	} else {
-		decayAdd = 0;
-		rateZero |= (1 << DECAY);
-	}
-}
-inline void Operator::UpdateRelease( const Chip* chip ) {
-	Bit8u rate = reg80 & 0xf;
-	if ( rate ) {
-		Bit8u val = (rate << 2) + ksr;
-		releaseAdd = chip->linearRates[ val ];
-		rateZero &= ~(1 << RELEASE);
-		if ( !(reg20 & MASK_SUSTAIN ) ) {
-			rateZero &= ~( 1 << SUSTAIN );
-		}
-	} else {
-		rateZero |= (1 << RELEASE);
-		releaseAdd = 0;
-		if ( !(reg20 & MASK_SUSTAIN ) ) {
-			rateZero |= ( 1 << SUSTAIN );
-		}
-	}
-}
-
-inline void Operator::UpdateAttenuation( ) {
-	Bit8u kslBase = (Bit8u)((chanData >> SHIFT_KSLBASE) & 0xff);
-	Bit32u tl = reg40 & 0x3f;
-	Bit8u kslShift = KslShiftTable[ reg40 >> 6 ];
-	//Make sure the attenuation goes to the right bits
-	totalLevel = tl << ( ENV_BITS - 7 );	//Total level goes 2 bits below max
-	totalLevel += ( kslBase << ENV_EXTRA ) >> kslShift;
-}
-
-void Operator::UpdateFrequency(  ) {
-	Bit32u freq = chanData & (( 1 << 10 ) - 1);
-	Bit32u block = (chanData >> 10) & 0xff;
-#ifdef WAVE_PRECISION
-	block = 7 - block;
-	waveAdd = ( freq * freqMul ) >> block;
-#else
-	waveAdd = ( freq << block ) * freqMul;
-#endif
-	if ( reg20 & MASK_VIBRATO ) {
-		vibStrength = (Bit8u)(freq >> 7);
-
-#ifdef WAVE_PRECISION
-		vibrato = ( vibStrength * freqMul ) >> block;
-#else
-		vibrato = ( vibStrength << block ) * freqMul;
-#endif
-	} else {
-		vibStrength = 0;
-		vibrato = 0;
-	}
-}
-
-void Operator::UpdateRates( const Chip* chip ) {
-	//Mame seems to reverse this where enabling ksr actually lowers
-	//the rate, but pdf manuals says otherwise?
-	Bit8u newKsr = (Bit8u)((chanData >> SHIFT_KEYCODE) & 0xff);
-	if ( !( reg20 & MASK_KSR ) ) {
-		newKsr >>= 2;
-	}
-	if ( ksr == newKsr )
-		return;
-	ksr = newKsr;
-	UpdateAttack( chip );
-	UpdateDecay( chip );
-	UpdateRelease( chip );
-}
-
-INLINE Bit32s Operator::RateForward( Bit32u add ) {
-	rateIndex += add;
-	Bit32s ret = rateIndex >> RATE_SH;
-	rateIndex = rateIndex & RATE_MASK;
-	return ret;
-}
-
-template< Operator::State yes>
-Bits Operator::TemplateVolume(  ) {
-	Bit32s vol = volume;
-	Bit32s change;
-	switch ( yes ) {
-	case OFF:
-		return ENV_MAX;
-	case ATTACK:
-		change = RateForward( attackAdd );
-		if ( !change )
-			return vol;
-		vol += ( (~vol) * change ) >> 3;
-		if ( vol < ENV_MIN ) {
-			volume = ENV_MIN;
-			rateIndex = 0;
-			SetState( DECAY );
-			return ENV_MIN;
-		}
-		break;
-	case DECAY:
-		vol += RateForward( decayAdd );
-		if ( GCC_UNLIKELY(vol >= sustainLevel) ) {
-			//Check if we didn't overshoot max attenuation, then just go off
-			if ( GCC_UNLIKELY(vol >= ENV_MAX) ) {
-				volume = ENV_MAX;
-				SetState( OFF );
-				return ENV_MAX;
-			}
-			//Continue as sustain
-			rateIndex = 0;
-			SetState( SUSTAIN );
-		}
-		break;
-	case SUSTAIN:
-		if ( reg20 & MASK_SUSTAIN ) {
-			return vol;
-		}
-		//In sustain phase, but not sustaining, do regular release
-	case RELEASE:
-		vol += RateForward( releaseAdd );
-		if ( GCC_UNLIKELY(vol >= ENV_MAX) ) {
-			volume = ENV_MAX;
-			SetState( OFF );
-			return ENV_MAX;
-		}
-		break;
-	}
-	volume = vol;
-	return vol;
-}
-
-static const VolumeHandler VolumeHandlerTable[5] = {
-	&Operator::TemplateVolume< Operator::OFF >,
-	&Operator::TemplateVolume< Operator::RELEASE >,
-	&Operator::TemplateVolume< Operator::SUSTAIN >,
-	&Operator::TemplateVolume< Operator::DECAY >,
-	&Operator::TemplateVolume< Operator::ATTACK >
-};
-
-INLINE Bitu Operator::ForwardVolume() {
-	return currentLevel + (this->*volHandler)();
-}
-
-
-INLINE Bitu Operator::ForwardWave() {
-	waveIndex += waveCurrent;
-	return waveIndex >> WAVE_SH;
-}
-
-void Operator::Write20( const Chip* chip, Bit8u val ) {
-	Bit8u change = (reg20 ^ val );
-	if ( !change )
-		return;
-	reg20 = val;
-	//Shift the tremolo bit over the entire register, saved a branch, YES!
-	tremoloMask = (Bit8s)(val) >> 7;
-	tremoloMask &= ~(( 1 << ENV_EXTRA ) -1);
-	//Update specific features based on changes
-	if ( change & MASK_KSR ) {
-		UpdateRates( chip );
-	}
-	//With sustain enable the volume doesn't change
-	if ( reg20 & MASK_SUSTAIN || ( !releaseAdd ) ) {
-		rateZero |= ( 1 << SUSTAIN );
-	} else {
-		rateZero &= ~( 1 << SUSTAIN );
-	}
-	//Frequency multiplier or vibrato changed
-	if ( change & (0xf | MASK_VIBRATO) ) {
-		freqMul = chip->freqMul[ val & 0xf ];
-		UpdateFrequency();
-	}
-}
-
-void Operator::Write40( const Chip* /*chip*/, Bit8u val ) {
-	if (!(reg40 ^ val ))
-		return;
-	reg40 = val;
-	UpdateAttenuation( );
-}
-
-void Operator::Write60( const Chip* chip, Bit8u val ) {
-	Bit8u change = reg60 ^ val;
-	reg60 = val;
-	if ( change & 0x0f ) {
-		UpdateDecay( chip );
-	}
-	if ( change & 0xf0 ) {
-		UpdateAttack( chip );
-	}
-}
-
-void Operator::Write80( const Chip* chip, Bit8u val ) {
-	Bit8u change = (reg80 ^ val );
-	if ( !change )
-		return;
-	reg80 = val;
-	Bit8u sustain = val >> 4;
-	//Turn 0xf into 0x1f
-	sustain |= ( sustain + 1) & 0x10;
-	sustainLevel = sustain << ( ENV_BITS - 5 );
-	if ( change & 0x0f ) {
-		UpdateRelease( chip );
-	}
-}
-
-void Operator::WriteE0( const Chip* chip, Bit8u val ) {
-	if ( !(regE0 ^ val) )
-		return;
-	//in opl3 mode you can always selet 7 waveforms regardless of waveformselect
-	Bit8u waveForm = val & ( ( 0x3 & chip->waveFormMask ) | (0x7 & chip->opl3Active ) );
-	regE0 = val;
-#if ( DBOPL_WAVE == WAVE_HANDLER )
-	waveHandler = WaveHandlerTable[ waveForm ];
-#else
-	waveBase = WaveTable + WaveBaseTable[ waveForm ];
-	waveStart = WaveStartTable[ waveForm ] << WAVE_SH;
-	waveMask = WaveMaskTable[ waveForm ];
-#endif
-}
-
-INLINE void Operator::SetState( Bit8u s ) {
-	state = s;
-	volHandler = VolumeHandlerTable[ s ];
-}
-
-INLINE bool Operator::Silent() const {
-	if ( !ENV_SILENT( totalLevel + volume ) )
-		return false;
-	if ( !(rateZero & ( 1 << state ) ) )
-		return false;
-	return true;
-}
-
-INLINE void Operator::Prepare( const Chip* chip )  {
-	currentLevel = totalLevel + (chip->tremoloValue & tremoloMask);
-	waveCurrent = waveAdd;
-	if ( vibStrength >> chip->vibratoShift ) {
-		Bit32s add = vibrato >> chip->vibratoShift;
-		//Sign extend over the shift value
-		Bit32s neg = chip->vibratoSign;
-		//Negate the add with -1 or 0
-		add = ( add ^ neg ) - neg;
-		waveCurrent += add;
-	}
-}
-
-void Operator::KeyOn( Bit8u mask ) {
-	if ( !keyOn ) {
-		//Restart the frequency generator
-#if ( DBOPL_WAVE > WAVE_HANDLER )
-		waveIndex = waveStart;
-#else
-		waveIndex = 0;
-#endif
-		rateIndex = 0;
-		SetState( ATTACK );
-	}
-	keyOn |= mask;
-}
-
-void Operator::KeyOff( Bit8u mask ) {
-	keyOn &= ~mask;
-	if ( !keyOn ) {
-		if ( state != OFF ) {
-			SetState( RELEASE );
-		}
-	}
-}
-
-INLINE Bits Operator::GetWave( Bitu index, Bitu vol ) {
-#if ( DBOPL_WAVE == WAVE_HANDLER )
-	return waveHandler( index, vol << ( 3 - ENV_EXTRA ) );
-#elif ( DBOPL_WAVE == WAVE_TABLEMUL )
-	return (waveBase[ index & waveMask ] * MulTable[ vol >> ENV_EXTRA ]) >> MUL_SH;
-#elif ( DBOPL_WAVE == WAVE_TABLELOG )
-	Bit32s wave = waveBase[ index & waveMask ];
-	Bit32u total = ( wave & 0x7fff ) + ( vol << ( 3 - ENV_EXTRA ) );
-	Bit32s sig = ExpTable[ total & 0xff ];
-	Bit32u exp = total >> 8;
-	Bit32s neg = wave >> 16;
-	return ((sig ^ neg) - neg) >> exp;
-#else
-#error "No valid wave routine"
-#endif
-}
-
-INLINE Bits Operator::GetSample( Bits modulation ) {
-	Bitu vol = ForwardVolume();
-	if ( ENV_SILENT( vol ) ) {
-		//Simply forward the wave
-		waveIndex += waveCurrent;
-		return 0;
-	} else {
-		Bitu index = ForwardWave();
-		index += modulation;
-		return GetWave( index, vol );
-	}
-}
-
-Operator::Operator() {
-	chanData = 0;
-	freqMul = 0;
-	waveIndex = 0;
-	waveAdd = 0;
-	waveCurrent = 0;
-	keyOn = 0;
-	ksr = 0;
-	reg20 = 0;
-	reg40 = 0;
-	reg60 = 0;
-	reg80 = 0;
-	regE0 = 0;
-	SetState( OFF );
-	rateZero = (1 << OFF);
-	sustainLevel = ENV_MAX;
-	currentLevel = ENV_MAX;
-	totalLevel = ENV_MAX;
-	volume = ENV_MAX;
-	releaseAdd = 0;
-}
-
-/*
-	Channel
-*/
-
-Channel::Channel() {
-	old[0] = old[1] = 0;
-	chanData = 0;
-	regB0 = 0;
-	regC0 = 0;
-	maskLeft = -1;
-	maskRight = -1;
-	feedback = 31;
-	fourMask = 0;
-	synthHandler = &Channel::BlockTemplate< sm2FM >;
-}
-
-void Channel::SetChanData( const Chip* chip, Bit32u data ) {
-	Bit32u change = chanData ^ data;
-	chanData = data;
-	Op( 0 )->chanData = data;
-	Op( 1 )->chanData = data;
-	//Since a frequency update triggered this, always update frequency
-	Op( 0 )->UpdateFrequency();
-	Op( 1 )->UpdateFrequency();
-	if ( change & ( 0xff << SHIFT_KSLBASE ) ) {
-		Op( 0 )->UpdateAttenuation();
-		Op( 1 )->UpdateAttenuation();
-	}
-	if ( change & ( 0xff << SHIFT_KEYCODE ) ) {
-		Op( 0 )->UpdateRates( chip );
-		Op( 1 )->UpdateRates( chip );
-	}
-}
-
-void Channel::UpdateFrequency( const Chip* chip, Bit8u fourOp ) {
-	//Extrace the frequency bits
-	Bit32u data = chanData & 0xffff;
-	Bit32u kslBase = KslTable[ data >> 6 ];
-	Bit32u keyCode = ( data & 0x1c00) >> 9;
-	if ( chip->reg08 & 0x40 ) {
-		keyCode |= ( data & 0x100)>>8;	/* notesel == 1 */
-	} else {
-		keyCode |= ( data & 0x200)>>9;	/* notesel == 0 */
-	}
-	//Add the keycode and ksl into the highest bits of chanData
-	data |= (keyCode << SHIFT_KEYCODE) | ( kslBase << SHIFT_KSLBASE );
-	( this + 0 )->SetChanData( chip, data );
-	if ( fourOp & 0x3f ) {
-		( this + 1 )->SetChanData( chip, data );
-	}
-}
-
-void Channel::WriteA0( const Chip* chip, Bit8u val ) {
-	Bit8u fourOp = chip->reg104 & chip->opl3Active & fourMask;
-	//Don't handle writes to silent fourop channels
-	if ( fourOp > 0x80 )
-		return;
-	Bit32u change = (chanData ^ val ) & 0xff;
-	if ( change ) {
-		chanData ^= change;
-		UpdateFrequency( chip, fourOp );
-	}
-}
-
-void Channel::WriteB0( const Chip* chip, Bit8u val ) {
-	Bit8u fourOp = chip->reg104 & chip->opl3Active & fourMask;
-	//Don't handle writes to silent fourop channels
-	if ( fourOp > 0x80 )
-		return;
-	Bitu change = (chanData ^ ( val << 8 ) ) & 0x1f00;
-	if ( change ) {
-		chanData ^= change;
-		UpdateFrequency( chip, fourOp );
-	}
-	//Check for a change in the keyon/off state
-	if ( !(( val ^ regB0) & 0x20))
-		return;
-	regB0 = val;
-	if ( val & 0x20 ) {
-		Op(0)->KeyOn( 0x1 );
-		Op(1)->KeyOn( 0x1 );
-		if ( fourOp & 0x3f ) {
-			( this + 1 )->Op(0)->KeyOn( 1 );
-			( this + 1 )->Op(1)->KeyOn( 1 );
-		}
-	} else {
-		Op(0)->KeyOff( 0x1 );
-		Op(1)->KeyOff( 0x1 );
-		if ( fourOp & 0x3f ) {
-			( this + 1 )->Op(0)->KeyOff( 1 );
-			( this + 1 )->Op(1)->KeyOff( 1 );
-		}
-	}
-}
-
-void Channel::WriteC0( const Chip* chip, Bit8u val ) {
-	Bit8u change = val ^ regC0;
-	if ( !change )
-		return;
-	regC0 = val;
-	feedback = ( val >> 1 ) & 7;
-	if ( feedback ) {
-		//We shift the input to the right 10 bit wave index value
-		feedback = 9 - feedback;
-	} else {
-		feedback = 31;
-	}
-	//Select the new synth mode
-	if ( chip->opl3Active ) {
-		//4-op mode enabled for this channel
-		if ( (chip->reg104 & fourMask) & 0x3f ) {
-			Channel* chan0, *chan1;
-			//Check if it's the 2nd channel in a 4-op
-			if ( !(fourMask & 0x80 ) ) {
-				chan0 = this;
-				chan1 = this + 1;
-			} else {
-				chan0 = this - 1;
-				chan1 = this;
-			}
-
-			Bit8u synth = ( (chan0->regC0 & 1) << 0 )| (( chan1->regC0 & 1) << 1 );
-			switch ( synth ) {
-			case 0:
-				chan0->synthHandler = &Channel::BlockTemplate< sm3FMFM >;
-				break;
-			case 1:
-				chan0->synthHandler = &Channel::BlockTemplate< sm3AMFM >;
-				break;
-			case 2:
-				chan0->synthHandler = &Channel::BlockTemplate< sm3FMAM >;
-				break;
-			case 3:
-				chan0->synthHandler = &Channel::BlockTemplate< sm3AMAM >;
-				break;
-			}
-		//Disable updating percussion channels
-		} else if ((fourMask & 0x40) && ( chip->regBD & 0x20) ) {
-
-		//Regular dual op, am or fm
-		} else if ( val & 1 ) {
-			synthHandler = &Channel::BlockTemplate< sm3AM >;
-		} else {
-			synthHandler = &Channel::BlockTemplate< sm3FM >;
-		}
-		maskLeft = ( val & 0x10 ) ? -1 : 0;
-		maskRight = ( val & 0x20 ) ? -1 : 0;
-	//opl2 active
-	} else {
-		//Disable updating percussion channels
-		if ( (fourMask & 0x40) && ( chip->regBD & 0x20 ) ) {
-
-		//Regular dual op, am or fm
-		} else if ( val & 1 ) {
-			synthHandler = &Channel::BlockTemplate< sm2AM >;
-		} else {
-			synthHandler = &Channel::BlockTemplate< sm2FM >;
-		}
-	}
-}
-
-void Channel::ResetC0( const Chip* chip ) {
-	Bit8u val = regC0;
-	regC0 ^= 0xff;
-	WriteC0( chip, val );
-}
-
-template< bool opl3Mode>
-INLINE void Channel::GeneratePercussion( Chip* chip, Bit32s* output ) {
-	Channel* chan = this;
-
-	//BassDrum
-	Bit32s mod = (Bit32u)((old[0] + old[1])) >> feedback;
-	old[0] = old[1];
-	old[1] = Op(0)->GetSample( mod );
-
-	//When bassdrum is in AM mode first operator is ignoed
-	if ( chan->regC0 & 1 ) {
-		mod = 0;
-	} else {
-		mod = old[0];
-	}
-	Bit32s sample = Op(1)->GetSample( mod );
-
-
-	//Precalculate stuff used by other outputs
-	Bit32u noiseBit = chip->ForwardNoise() & 0x1;
-	Bit32u c2 = Op(2)->ForwardWave();
-	Bit32u c5 = Op(5)->ForwardWave();
-	Bit32u phaseBit = (((c2 & 0x88) ^ ((c2<<5) & 0x80)) | ((c5 ^ (c5<<2)) & 0x20)) ? 0x02 : 0x00;
-
-	//Hi-Hat
-	Bit32u hhVol = Op(2)->ForwardVolume();
-	if ( !ENV_SILENT( hhVol ) ) {
-		Bit32u hhIndex = (phaseBit<<8) | (0x34 << ( phaseBit ^ (noiseBit << 1 )));
-		sample += Op(2)->GetWave( hhIndex, hhVol );
-	}
-	//Snare Drum
-	Bit32u sdVol = Op(3)->ForwardVolume();
-	if ( !ENV_SILENT( sdVol ) ) {
-		Bit32u sdIndex = ( 0x100 + (c2 & 0x100) ) ^ ( noiseBit << 8 );
-		sample += Op(3)->GetWave( sdIndex, sdVol );
-	}
-	//Tom-tom
-	sample += Op(4)->GetSample( 0 );
-
-	//Top-Cymbal
-	Bit32u tcVol = Op(5)->ForwardVolume();
-	if ( !ENV_SILENT( tcVol ) ) {
-		Bit32u tcIndex = (1 + phaseBit) << 8;
-		sample += Op(5)->GetWave( tcIndex, tcVol );
-	}
-	sample <<= 1;
-	if ( opl3Mode ) {
-		output[0] += sample;
-		output[1] += sample;
-	} else {
-		output[0] += sample;
-	}
-}
-
-template<SynthMode mode>
-Channel* Channel::BlockTemplate( Chip* chip, Bit32u samples, Bit32s* output ) {
-	switch( mode ) {
-	case sm2AM:
-	case sm3AM:
-		if ( Op(0)->Silent() && Op(1)->Silent() ) {
-			old[0] = old[1] = 0;
-			return (this + 1);
-		}
-		break;
-	case sm2FM:
-	case sm3FM:
-		if ( Op(1)->Silent() ) {
-			old[0] = old[1] = 0;
-			return (this + 1);
-		}
-		break;
-	case sm3FMFM:
-		if ( Op(3)->Silent() ) {
-			old[0] = old[1] = 0;
-			return (this + 2);
-		}
-		break;
-	case sm3AMFM:
-		if ( Op(0)->Silent() && Op(3)->Silent() ) {
-			old[0] = old[1] = 0;
-			return (this + 2);
-		}
-		break;
-	case sm3FMAM:
-		if ( Op(1)->Silent() && Op(3)->Silent() ) {
-			old[0] = old[1] = 0;
-			return (this + 2);
-		}
-		break;
-	case sm3AMAM:
-		if ( Op(0)->Silent() && Op(2)->Silent() && Op(3)->Silent() ) {
-			old[0] = old[1] = 0;
-			return (this + 2);
-		}
-		break;
-	case sm2Percussion:
-		// This case was not handled in the DOSBox code either
-		// thus we leave this blank.
-		// TODO: Consider checking this.
-		break;
-	case sm3Percussion:
-		// This case was not handled in the DOSBox code either
-		// thus we leave this blank.
-		// TODO: Consider checking this.
-		break;
-	case sm4Start:
-		// This case was not handled in the DOSBox code either
-		// thus we leave this blank.
-		// TODO: Consider checking this.
-		break;
-	case sm6Start:
-		// This case was not handled in the DOSBox code either
-		// thus we leave this blank.
-		// TODO: Consider checking this.
-		break;
-	}
-	//Init the operators with the the current vibrato and tremolo values
-	Op( 0 )->Prepare( chip );
-	Op( 1 )->Prepare( chip );
-	if ( mode > sm4Start ) {
-		Op( 2 )->Prepare( chip );
-		Op( 3 )->Prepare( chip );
-	}
-	if ( mode > sm6Start ) {
-		Op( 4 )->Prepare( chip );
-		Op( 5 )->Prepare( chip );
-	}
-	for ( Bitu i = 0; i < samples; i++ ) {
-		//Early out for percussion handlers
-		if ( mode == sm2Percussion ) {
-			GeneratePercussion<false>( chip, output + i );
-			continue;	//Prevent some unitialized value bitching
-		} else if ( mode == sm3Percussion ) {
-			GeneratePercussion<true>( chip, output + i * 2 );
-			continue;	//Prevent some unitialized value bitching
-		}
-
-		//Do unsigned shift so we can shift out all bits but still stay in 10 bit range otherwise
-		Bit32s mod = (Bit32u)((old[0] + old[1])) >> feedback;
-		old[0] = old[1];
-		old[1] = Op(0)->GetSample( mod );
-		Bit32s sample;
-		Bit32s out0 = old[0];
-		if ( mode == sm2AM || mode == sm3AM ) {
-			sample = out0 + Op(1)->GetSample( 0 );
-		} else if ( mode == sm2FM || mode == sm3FM ) {
-			sample = Op(1)->GetSample( out0 );
-		} else if ( mode == sm3FMFM ) {
-			Bits next = Op(1)->GetSample( out0 );
-			next = Op(2)->GetSample( next );
-			sample = Op(3)->GetSample( next );
-		} else if ( mode == sm3AMFM ) {
-			sample = out0;
-			Bits next = Op(1)->GetSample( 0 );
-			next = Op(2)->GetSample( next );
-			sample += Op(3)->GetSample( next );
-		} else if ( mode == sm3FMAM ) {
-			sample = Op(1)->GetSample( out0 );
-			Bits next = Op(2)->GetSample( 0 );
-			sample += Op(3)->GetSample( next );
-		} else if ( mode == sm3AMAM ) {
-			sample = out0;
-			Bits next = Op(1)->GetSample( 0 );
-			sample += Op(2)->GetSample( next );
-			sample += Op(3)->GetSample( 0 );
-		}
-		switch( mode ) {
-		case sm2AM:
-		case sm2FM:
-			output[ i ] += sample;
-			break;
-		case sm3AM:
-		case sm3FM:
-		case sm3FMFM:
-		case sm3AMFM:
-		case sm3FMAM:
-		case sm3AMAM:
-			output[ i * 2 + 0 ] += sample & maskLeft;
-			output[ i * 2 + 1 ] += sample & maskRight;
-			break;
-		case sm2Percussion:
-			// This case was not handled in the DOSBox code either
-			// thus we leave this blank.
-			// TODO: Consider checking this.
-			break;
-		case sm3Percussion:
-			// This case was not handled in the DOSBox code either
-			// thus we leave this blank.
-			// TODO: Consider checking this.
-			break;
-		case sm4Start:
-			// This case was not handled in the DOSBox code either
-			// thus we leave this blank.
-			// TODO: Consider checking this.
-			break;
-		case sm6Start:
-			// This case was not handled in the DOSBox code either
-			// thus we leave this blank.
-			// TODO: Consider checking this.
-			break;
-		}
-	}
-	switch( mode ) {
-	case sm2AM:
-	case sm2FM:
-	case sm3AM:
-	case sm3FM:
-		return ( this + 1 );
-	case sm3FMFM:
-	case sm3AMFM:
-	case sm3FMAM:
-	case sm3AMAM:
-		return( this + 2 );
-	case sm2Percussion:
-	case sm3Percussion:
-		return( this + 3 );
-	case sm4Start:
-		// This case was not handled in the DOSBox code either
-		// thus we leave this blank.
-		// TODO: Consider checking this.
-		break;
-	case sm6Start:
-		// This case was not handled in the DOSBox code either
-		// thus we leave this blank.
-		// TODO: Consider checking this.
-		break;
-	}
-	return 0;
-}
-
-/*
-	Chip
-*/
-
-Chip::Chip() {
-	reg08 = 0;
-	reg04 = 0;
-	regBD = 0;
-	reg104 = 0;
-	opl3Active = 0;
-}
-
-INLINE Bit32u Chip::ForwardNoise() {
-	noiseCounter += noiseAdd;
-	Bitu count = noiseCounter >> LFO_SH;
-	noiseCounter &= WAVE_MASK;
-	for ( ; count > 0; --count ) {
-		//Noise calculation from mame
-		noiseValue ^= ( 0x800302 ) & ( 0 - (noiseValue & 1 ) );
-		noiseValue >>= 1;
-	}
-	return noiseValue;
-}
-
-INLINE Bit32u Chip::ForwardLFO( Bit32u samples ) {
-	//Current vibrato value, runs 4x slower than tremolo
-	vibratoSign = ( VibratoTable[ vibratoIndex >> 2] ) >> 7;
-	vibratoShift = ( VibratoTable[ vibratoIndex >> 2] & 7) + vibratoStrength;
-	tremoloValue = TremoloTable[ tremoloIndex ] >> tremoloStrength;
-
-	//Check hom many samples there can be done before the value changes
-	Bit32u todo = LFO_MAX - lfoCounter;
-	Bit32u count = (todo + lfoAdd - 1) / lfoAdd;
-	if ( count > samples ) {
-		count = samples;
-		lfoCounter += count * lfoAdd;
-	} else {
-		lfoCounter += count * lfoAdd;
-		lfoCounter &= (LFO_MAX - 1);
-		//Maximum of 7 vibrato value * 4
-		vibratoIndex = ( vibratoIndex + 1 ) & 31;
-		//Clip tremolo to the the table size
-		if ( tremoloIndex + 1 < TREMOLO_TABLE  )
-			++tremoloIndex;
-		else
-			tremoloIndex = 0;
-	}
-	return count;
-}
-
-
-void Chip::WriteBD( Bit8u val ) {
-	Bit8u change = regBD ^ val;
-	if ( !change )
-		return;
-	regBD = val;
-	//TODO could do this with shift and xor?
-	vibratoStrength = (val & 0x40) ? 0x00 : 0x01;
-	tremoloStrength = (val & 0x80) ? 0x00 : 0x02;
-	if ( val & 0x20 ) {
-		//Drum was just enabled, make sure channel 6 has the right synth
-		if ( change & 0x20 ) {
-			if ( opl3Active ) {
-				chan[6].synthHandler = &Channel::BlockTemplate< sm3Percussion >;
-			} else {
-				chan[6].synthHandler = &Channel::BlockTemplate< sm2Percussion >;
-			}
-		}
-		//Bass Drum
-		if ( val & 0x10 ) {
-			chan[6].op[0].KeyOn( 0x2 );
-			chan[6].op[1].KeyOn( 0x2 );
-		} else {
-			chan[6].op[0].KeyOff( 0x2 );
-			chan[6].op[1].KeyOff( 0x2 );
-		}
-		//Hi-Hat
-		if ( val & 0x1 ) {
-			chan[7].op[0].KeyOn( 0x2 );
-		} else {
-			chan[7].op[0].KeyOff( 0x2 );
-		}
-		//Snare
-		if ( val & 0x8 ) {
-			chan[7].op[1].KeyOn( 0x2 );
-		} else {
-			chan[7].op[1].KeyOff( 0x2 );
-		}
-		//Tom-Tom
-		if ( val & 0x4 ) {
-			chan[8].op[0].KeyOn( 0x2 );
-		} else {
-			chan[8].op[0].KeyOff( 0x2 );
-		}
-		//Top Cymbal
-		if ( val & 0x2 ) {
-			chan[8].op[1].KeyOn( 0x2 );
-		} else {
-			chan[8].op[1].KeyOff( 0x2 );
-		}
-	//Toggle keyoffs when we turn off the percussion
-	} else if ( change & 0x20 ) {
-		//Trigger a reset to setup the original synth handler
-		chan[6].ResetC0( this );
-		chan[6].op[0].KeyOff( 0x2 );
-		chan[6].op[1].KeyOff( 0x2 );
-		chan[7].op[0].KeyOff( 0x2 );
-		chan[7].op[1].KeyOff( 0x2 );
-		chan[8].op[0].KeyOff( 0x2 );
-		chan[8].op[1].KeyOff( 0x2 );
-	}
-}
-
-
-#define REGOP( _FUNC_ )															\
-	index = ( ( reg >> 3) & 0x20 ) | ( reg & 0x1f );								\
-	if ( OpOffsetTable[ index ] ) {													\
-		Operator* regOp = (Operator*)( ((char *)this ) + OpOffsetTable[ index ] );	\
-		regOp->_FUNC_( this, val );													\
-	}
-
-#define REGCHAN( _FUNC_ )																\
-	index = ( ( reg >> 4) & 0x10 ) | ( reg & 0xf );										\
-	if ( ChanOffsetTable[ index ] ) {													\
-		Channel* regChan = (Channel*)( ((char *)this ) + ChanOffsetTable[ index ] );	\
-		regChan->_FUNC_( this, val );													\
-	}
-
-void Chip::WriteReg( Bit32u reg, Bit8u val ) {
-	Bitu index;
-	switch ( (reg & 0xf0) >> 4 ) {
-	case 0x00 >> 4:
-		if ( reg == 0x01 ) {
-			waveFormMask = ( val & 0x20 ) ? 0x7 : 0x0;
-		} else if ( reg == 0x104 ) {
-			//Only detect changes in lowest 6 bits
-			if ( !((reg104 ^ val) & 0x3f) )
-				return;
-			//Always keep the highest bit enabled, for checking > 0x80
-			reg104 = 0x80 | ( val & 0x3f );
-		} else if ( reg == 0x105 ) {
-			//MAME says the real opl3 doesn't reset anything on opl3 disable/enable till the next write in another register
-			if ( !((opl3Active ^ val) & 1 ) )
-				return;
-			opl3Active = ( val & 1 ) ? 0xff : 0;
-			//Update the 0xc0 register for all channels to signal the switch to mono/stereo handlers
-			for ( int i = 0; i < 18;i++ ) {
-				chan[i].ResetC0( this );
-			}
-		} else if ( reg == 0x08 ) {
-			reg08 = val;
-		}
-	case 0x10 >> 4:
-		break;
-	case 0x20 >> 4:
-	case 0x30 >> 4:
-		REGOP( Write20 );
-		break;
-	case 0x40 >> 4:
-	case 0x50 >> 4:
-		REGOP( Write40 );
-		break;
-	case 0x60 >> 4:
-	case 0x70 >> 4:
-		REGOP( Write60 );
-		break;
-	case 0x80 >> 4:
-	case 0x90 >> 4:
-		REGOP( Write80 );
-		break;
-	case 0xa0 >> 4:
-		REGCHAN( WriteA0 );
-		break;
-	case 0xb0 >> 4:
-		if ( reg == 0xbd ) {
-			WriteBD( val );
-		} else {
-			REGCHAN( WriteB0 );
-		}
-		break;
-	case 0xc0 >> 4:
-		REGCHAN( WriteC0 );
-	case 0xd0 >> 4:
-		break;
-	case 0xe0 >> 4:
-	case 0xf0 >> 4:
-		REGOP( WriteE0 );
-		break;
-	}
-}
-
-
-Bit32u Chip::WriteAddr( Bit32u port, Bit8u val ) {
-	switch ( port & 3 ) {
-	case 0:
-		return val;
-	case 2:
-		if ( opl3Active || (val == 0x05) )
-			return 0x100 | val;
-		else
-			return val;
-	}
-	return 0;
-}
-
-void Chip::GenerateBlock2( Bitu total, Bit32s* output ) {
-	while ( total > 0 ) {
-		Bit32u samples = ForwardLFO( total );
-		memset(output, 0, sizeof(Bit32s) * samples);
-		int count = 0;
-		for( Channel* ch = chan; ch < chan + 9; ) {
-			count++;
-			ch = (ch->*(ch->synthHandler))( this, samples, output );
-		}
-		total -= samples;
-		output += samples;
-	}
-}
-
-void Chip::GenerateBlock3( Bitu total, Bit32s* output  ) {
-	while ( total > 0 ) {
-		Bit32u samples = ForwardLFO( total );
-		memset(output, 0, sizeof(Bit32s) * 2 * samples);
-		int count = 0;
-		for( Channel* ch = chan; ch < chan + 18; ) {
-			count++;
-			ch = (ch->*(ch->synthHandler))( this, samples, output );
-		}
-		total -= samples;
-		output += samples * 2;
-	}
-}
-
-void Chip::Setup( Bit32u rate ) {
-	double scale = OPLRATE / (double)rate;
-
-	//Noise counter is run at the same precision as general waves
-	noiseAdd = (Bit32u)( 0.5 + scale * ( 1 << LFO_SH ) );
-	noiseCounter = 0;
-	noiseValue = 1;	//Make sure it triggers the noise xor the first time
-	//The low frequency oscillation counter
-	//Every time his overflows vibrato and tremoloindex are increased
-	lfoAdd = (Bit32u)( 0.5 + scale * ( 1 << LFO_SH ) );
-	lfoCounter = 0;
-	vibratoIndex = 0;
-	tremoloIndex = 0;
-
-	//With higher octave this gets shifted up
-	//-1 since the freqCreateTable = *2
-#ifdef WAVE_PRECISION
-	double freqScale = ( 1 << 7 ) * scale * ( 1 << ( WAVE_SH - 1 - 10));
-	for ( int i = 0; i < 16; i++ ) {
-		freqMul[i] = (Bit32u)( 0.5 + freqScale * FreqCreateTable[ i ] );
-	}
-#else
-	Bit32u freqScale = (Bit32u)( 0.5 + scale * ( 1 << ( WAVE_SH - 1 - 10)));
-	for ( int i = 0; i < 16; i++ ) {
-		freqMul[i] = freqScale * FreqCreateTable[ i ];
-	}
-#endif
-
-	//-3 since the real envelope takes 8 steps to reach the single value we supply
-	for ( Bit8u i = 0; i < 76; i++ ) {
-		Bit8u index, shift;
-		EnvelopeSelect( i, index, shift );
-		linearRates[i] = (Bit32u)( scale * (EnvelopeIncreaseTable[ index ] << ( RATE_SH + ENV_EXTRA - shift - 3 )));
-	}
-	//Generate the best matching attack rate
-	for ( Bit8u i = 0; i < 62; i++ ) {
-		Bit8u index, shift;
-		EnvelopeSelect( i, index, shift );
-		//Original amount of samples the attack would take
-		Bit32s original = (Bit32u)( (AttackSamplesTable[ index ] << shift) / scale);
-
-		Bit32s guessAdd = (Bit32u)( scale * (EnvelopeIncreaseTable[ index ] << ( RATE_SH - shift - 3 )));
-		Bit32s bestAdd = guessAdd;
-		Bit32u bestDiff = 1 << 30;
-		for( Bit32u passes = 0; passes < 16; passes ++ ) {
-			Bit32s volume = ENV_MAX;
-			Bit32s samples = 0;
-			Bit32u count = 0;
-			while ( volume > 0 && samples < original * 2 ) {
-				count += guessAdd;
-				Bit32s change = count >> RATE_SH;
-				count &= RATE_MASK;
-				if ( GCC_UNLIKELY(change) ) { // less than 1 %
-					volume += ( ~volume * change ) >> 3;
-				}
-				samples++;
-
-			}
-			Bit32s diff = original - samples;
-			Bit32u lDiff = labs( diff );
-			//Init last on first pass
-			if ( lDiff < bestDiff ) {
-				bestDiff = lDiff;
-				bestAdd = guessAdd;
-				if ( !bestDiff )
-					break;
-			}
-			//Below our target
-			if ( diff < 0 ) {
-				//Better than the last time
-				Bit32s mul = ((original - diff) << 12) / original;
-				guessAdd = ((guessAdd * mul) >> 12);
-				guessAdd++;
-			} else if ( diff > 0 ) {
-				Bit32s mul = ((original - diff) << 12) / original;
-				guessAdd = (guessAdd * mul) >> 12;
-				guessAdd--;
-			}
-		}
-		attackRates[i] = bestAdd;
-	}
-	for ( Bit8u i = 62; i < 76; i++ ) {
-		//This should provide instant volume maximizing
-		attackRates[i] = 8 << RATE_SH;
-	}
-	//Setup the channels with the correct four op flags
-	//Channels are accessed through a table so they appear linear here
-	chan[ 0].fourMask = 0x00 | ( 1 << 0 );
-	chan[ 1].fourMask = 0x80 | ( 1 << 0 );
-	chan[ 2].fourMask = 0x00 | ( 1 << 1 );
-	chan[ 3].fourMask = 0x80 | ( 1 << 1 );
-	chan[ 4].fourMask = 0x00 | ( 1 << 2 );
-	chan[ 5].fourMask = 0x80 | ( 1 << 2 );
-
-	chan[ 9].fourMask = 0x00 | ( 1 << 3 );
-	chan[10].fourMask = 0x80 | ( 1 << 3 );
-	chan[11].fourMask = 0x00 | ( 1 << 4 );
-	chan[12].fourMask = 0x80 | ( 1 << 4 );
-	chan[13].fourMask = 0x00 | ( 1 << 5 );
-	chan[14].fourMask = 0x80 | ( 1 << 5 );
-
-	//mark the percussion channels
-	chan[ 6].fourMask = 0x40;
-	chan[ 7].fourMask = 0x40;
-	chan[ 8].fourMask = 0x40;
-
-	//Clear Everything in opl3 mode
-	WriteReg( 0x105, 0x1 );
-	for ( int i = 0; i < 512; i++ ) {
-		if ( i == 0x105 )
-			continue;
-		WriteReg( i, 0xff );
-		WriteReg( i, 0x0 );
-	}
-	WriteReg( 0x105, 0x0 );
-	//Clear everything in opl2 mode
-	for ( int i = 0; i < 255; i++ ) {
-		WriteReg( i, 0xff );
-		WriteReg( i, 0x0 );
-	}
-}
-
-static bool doneTables = false;
-void InitTables( void ) {
-	if ( doneTables )
-		return;
-	doneTables = true;
-#if ( DBOPL_WAVE == WAVE_HANDLER ) || ( DBOPL_WAVE == WAVE_TABLELOG )
-	//Exponential volume table, same as the real adlib
-	for ( int i = 0; i < 256; i++ ) {
-		//Save them in reverse
-		ExpTable[i] = (int)( 0.5 + ( pow(2.0, ( 255 - i) * ( 1.0 /256 ) )-1) * 1024 );
-		ExpTable[i] += 1024; //or remove the -1 oh well :)
-		//Preshift to the left once so the final volume can shift to the right
-		ExpTable[i] *= 2;
-	}
-#endif
-#if ( DBOPL_WAVE == WAVE_HANDLER )
-	//Add 0.5 for the trunc rounding of the integer cast
-	//Do a PI sinetable instead of the original 0.5 PI
-	for ( int i = 0; i < 512; i++ ) {
-		SinTable[i] = (Bit16s)( 0.5 - log10( sin( (i + 0.5) * (PI / 512.0) ) ) / log10(2.0)*256 );
-	}
-#endif
-#if ( DBOPL_WAVE == WAVE_TABLEMUL )
-	//Multiplication based tables
-	for ( int i = 0; i < 384; i++ ) {
-		int s = i * 8;
-		//TODO maybe keep some of the precision errors of the original table?
-		double val = ( 0.5 + ( pow(2.0, -1.0 + ( 255 - s) * ( 1.0 /256 ) )) * ( 1 << MUL_SH ));
-		MulTable[i] = (Bit16u)(val);
-	}
-
-	//Sine Wave Base
-	for ( int i = 0; i < 512; i++ ) {
-		WaveTable[ 0x0200 + i ] = (Bit16s)(sin( (i + 0.5) * (PI / 512.0) ) * 4084);
-		WaveTable[ 0x0000 + i ] = -WaveTable[ 0x200 + i ];
-	}
-	//Exponential wave
-	for ( int i = 0; i < 256; i++ ) {
-		WaveTable[ 0x700 + i ] = (Bit16s)( 0.5 + ( pow(2.0, -1.0 + ( 255 - i * 8) * ( 1.0 /256 ) ) ) * 4085 );
-		WaveTable[ 0x6ff - i ] = -WaveTable[ 0x700 + i ];
-	}
-#endif
-#if ( DBOPL_WAVE == WAVE_TABLELOG )
-	//Sine Wave Base
-	for ( int i = 0; i < 512; i++ ) {
-		WaveTable[ 0x0200 + i ] = (Bit16s)( 0.5 - log10( sin( (i + 0.5) * (PI / 512.0) ) ) / log10(2.0)*256 );
-		WaveTable[ 0x0000 + i ] = ((Bit16s)0x8000) | WaveTable[ 0x200 + i];
-	}
-	//Exponential wave
-	for ( int i = 0; i < 256; i++ ) {
-		WaveTable[ 0x700 + i ] = i * 8;
-		WaveTable[ 0x6ff - i ] = ((Bit16s)0x8000) | i * 8;
-	}
-#endif
-
-	//	|    |//\\|____|WAV7|//__|/\  |____|/\/\|
-	//	|\\//|    |    |WAV7|    |  \/|    |    |
-	//	|06  |0126|27  |7   |3   |4   |4 5 |5   |
-
-#if (( DBOPL_WAVE == WAVE_TABLELOG ) || ( DBOPL_WAVE == WAVE_TABLEMUL ))
-	for ( int i = 0; i < 256; i++ ) {
-		//Fill silence gaps
-		WaveTable[ 0x400 + i ] = WaveTable[0];
-		WaveTable[ 0x500 + i ] = WaveTable[0];
-		WaveTable[ 0x900 + i ] = WaveTable[0];
-		WaveTable[ 0xc00 + i ] = WaveTable[0];
-		WaveTable[ 0xd00 + i ] = WaveTable[0];
-		//Replicate sines in other pieces
-		WaveTable[ 0x800 + i ] = WaveTable[ 0x200 + i ];
-		//double speed sines
-		WaveTable[ 0xa00 + i ] = WaveTable[ 0x200 + i * 2 ];
-		WaveTable[ 0xb00 + i ] = WaveTable[ 0x000 + i * 2 ];
-		WaveTable[ 0xe00 + i ] = WaveTable[ 0x200 + i * 2 ];
-		WaveTable[ 0xf00 + i ] = WaveTable[ 0x200 + i * 2 ];
-	}
-#endif
-
-	//Create the ksl table
-	for ( int oct = 0; oct < 8; oct++ ) {
-		int base = oct * 8;
-		for ( int i = 0; i < 16; i++ ) {
-			int val = base - KslCreateTable[i];
-			if ( val < 0 )
-				val = 0;
-			//*4 for the final range to match attenuation range
-			KslTable[ oct * 16 + i ] = val * 4;
-		}
-	}
-	//Create the Tremolo table, just increase and decrease a triangle wave
-	for ( Bit8u i = 0; i < TREMOLO_TABLE / 2; i++ ) {
-		Bit8u val = i << ENV_EXTRA;
-		TremoloTable[i] = val;
-		TremoloTable[TREMOLO_TABLE - 1 - i] = val;
-	}
-	//Create a table with offsets of the channels from the start of the chip
-	DBOPL::Chip* chip = 0;
-	for ( Bitu i = 0; i < 32; i++ ) {
-		Bitu index = i & 0xf;
-		if ( index >= 9 ) {
-			ChanOffsetTable[i] = 0;
-			continue;
-		}
-		//Make sure the four op channels follow eachother
-		if ( index < 6 ) {
-			index = (index % 3) * 2 + ( index / 3 );
-		}
-		//Add back the bits for highest ones
-		if ( i >= 16 )
-			index += 9;
-		Bitu blah = reinterpret_cast<size_t>( &(chip->chan[ index ]) );
-		ChanOffsetTable[i] = blah;
-	}
-	//Same for operators
-	for ( Bitu i = 0; i < 64; i++ ) {
-		if ( i % 8 >= 6 || ( (i / 8) % 4 == 3 ) ) {
-			OpOffsetTable[i] = 0;
-			continue;
-		}
-		Bitu chNum = (i / 8) * 3 + (i % 8) % 3;
-		//Make sure we use 16 and up for the 2nd range to match the chanoffset gap
-		if ( chNum >= 12 )
-			chNum += 16 - 12;
-		Bitu opNum = ( i % 8 ) / 3;
-		DBOPL::Channel* chan = 0;
-		Bitu blah = reinterpret_cast<size_t>( &(chan->op[opNum]) );
-		OpOffsetTable[i] = ChanOffsetTable[ chNum ] + blah;
-	}
-#if 0
-	//Stupid checks if table's are correct
-	for ( Bitu i = 0; i < 18; i++ ) {
-		Bit32u find = (Bit16u)( &(chip->chan[ i ]) );
-		for ( Bitu c = 0; c < 32; c++ ) {
-			if ( ChanOffsetTable[c] == find ) {
-				find = 0;
-				break;
-			}
-		}
-		if ( find ) {
-			find = find;
-		}
-	}
-	for ( Bitu i = 0; i < 36; i++ ) {
-		Bit32u find = (Bit16u)( &(chip->chan[ i / 2 ].op[i % 2]) );
-		for ( Bitu c = 0; c < 64; c++ ) {
-			if ( OpOffsetTable[c] == find ) {
-				find = 0;
-				break;
-			}
-		}
-		if ( find ) {
-			find = find;
-		}
-	}
-#endif
-}
-
-}		//Namespace DBOPL
-} // End of namespace DOSBox
-} // End of namespace OPL
-
-#endif // !DISABLE_DOSBOX_OPL
diff --git a/sound/softsynth/opl/dbopl.h b/sound/softsynth/opl/dbopl.h
deleted file mode 100644
index 87d1045fab..0000000000
--- a/sound/softsynth/opl/dbopl.h
+++ /dev/null
@@ -1,283 +0,0 @@
-/*
- *  Copyright (C) 2002-2010  The DOSBox Team
- *
- *  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., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
- */
-
-// Last synch with DOSBox SVN trunk r3556
-
-#ifndef SOUND_SOFTSYNTH_OPL_DBOPL_H
-#define SOUND_SOFTSYNTH_OPL_DBOPL_H
-
-#include "common/scummsys.h"
-
-#ifndef DISABLE_DOSBOX_OPL
-
-namespace OPL {
-namespace DOSBox {
-
-//Use 8 handlers based on a small logatirmic wavetabe and an exponential table for volume
-#define WAVE_HANDLER	10
-//Use a logarithmic wavetable with an exponential table for volume
-#define WAVE_TABLELOG	11
-//Use a linear wavetable with a multiply table for volume
-#define WAVE_TABLEMUL	12
-
-//Select the type of wave generator routine
-#define DBOPL_WAVE WAVE_TABLEMUL
-
-namespace DBOPL {
-
-// Type aliases for the DBOPL code
-typedef int Bits;
-typedef uint Bitu;
-
-typedef int8 Bit8s;
-typedef uint8 Bit8u;
-
-typedef int16 Bit16s;
-typedef uint16 Bit16u;
-
-typedef int32 Bit32s;
-typedef uint32 Bit32u;
-
-#define DB_FASTCALL
-#define GCC_UNLIKELY(x) (x)
-#define INLINE inline
-// -------------------------------
-
-struct Chip;
-struct Operator;
-struct Channel;
-
-#if (DBOPL_WAVE == WAVE_HANDLER)
-typedef Bits ( DB_FASTCALL *WaveHandler) ( Bitu i, Bitu volume );
-#endif
-
-typedef Bits ( DBOPL::Operator::*VolumeHandler) ( );
-typedef Channel* ( DBOPL::Channel::*SynthHandler) ( Chip* chip, Bit32u samples, Bit32s* output );
-
-//Different synth modes that can generate blocks of data
-typedef enum {
-	sm2AM,
-	sm2FM,
-	sm3AM,
-	sm3FM,
-	sm4Start,
-	sm3FMFM,
-	sm3AMFM,
-	sm3FMAM,
-	sm3AMAM,
-	sm6Start,
-	sm2Percussion,
-	sm3Percussion
-} SynthMode;
-
-//Shifts for the values contained in chandata variable
-enum {
-	SHIFT_KSLBASE = 16,
-	SHIFT_KEYCODE = 24
-};
-
-struct Operator {
-public:
-	//Masks for operator 20 values
-	enum {
-		MASK_KSR = 0x10,
-		MASK_SUSTAIN = 0x20,
-		MASK_VIBRATO = 0x40,
-		MASK_TREMOLO = 0x80
-	};
-
-	typedef enum {
-		OFF,
-		RELEASE,
-		SUSTAIN,
-		DECAY,
-		ATTACK
-	} State;
-
-	VolumeHandler volHandler;
-
-#if (DBOPL_WAVE == WAVE_HANDLER)
-	WaveHandler waveHandler;	//Routine that generate a wave
-#else
-	Bit16s* waveBase;
-	Bit32u waveMask;
-	Bit32u waveStart;
-#endif
-	Bit32u waveIndex;			//WAVE_BITS shifted counter of the frequency index
-	Bit32u waveAdd;				//The base frequency without vibrato
-	Bit32u waveCurrent;			//waveAdd + vibratao
-
-	Bit32u chanData;			//Frequency/octave and derived data coming from whatever channel controls this
-	Bit32u freqMul;				//Scale channel frequency with this, TODO maybe remove?
-	Bit32u vibrato;				//Scaled up vibrato strength
-	Bit32s sustainLevel;		//When stopping at sustain level stop here
-	Bit32s totalLevel;			//totalLevel is added to every generated volume
-	Bit32u currentLevel;		//totalLevel + tremolo
-	Bit32s volume;				//The currently active volume
-
-	Bit32u attackAdd;			//Timers for the different states of the envelope
-	Bit32u decayAdd;
-	Bit32u releaseAdd;
-	Bit32u rateIndex;			//Current position of the evenlope
-
-	Bit8u rateZero;				//Bits for the different states of the envelope having no changes
-	Bit8u keyOn;				//Bitmask of different values that can generate keyon
-	//Registers, also used to check for changes
-	Bit8u reg20, reg40, reg60, reg80, regE0;
-	//Active part of the envelope we're in
-	Bit8u state;
-	//0xff when tremolo is enabled
-	Bit8u tremoloMask;
-	//Strength of the vibrato
-	Bit8u vibStrength;
-	//Keep track of the calculated KSR so we can check for changes
-	Bit8u ksr;
-private:
-	void SetState( Bit8u s );
-	void UpdateAttack( const Chip* chip );
-	void UpdateRelease( const Chip* chip );
-	void UpdateDecay( const Chip* chip );
-public:
-	void UpdateAttenuation();
-	void UpdateRates( const Chip* chip );
-	void UpdateFrequency( );
-
-	void Write20( const Chip* chip, Bit8u val );
-	void Write40( const Chip* chip, Bit8u val );
-	void Write60( const Chip* chip, Bit8u val );
-	void Write80( const Chip* chip, Bit8u val );
-	void WriteE0( const Chip* chip, Bit8u val );
-
-	bool Silent() const;
-	void Prepare( const Chip* chip );
-
-	void KeyOn( Bit8u mask);
-	void KeyOff( Bit8u mask);
-
-	template< State state>
-	Bits TemplateVolume( );
-
-	Bit32s RateForward( Bit32u add );
-	Bitu ForwardWave();
-	Bitu ForwardVolume();
-
-	Bits GetSample( Bits modulation );
-	Bits GetWave( Bitu index, Bitu vol );
-public:
-	Operator();
-};
-
-struct Channel {
-	Operator op[2];
-	inline Operator* Op( Bitu index ) {
-		return &( ( this + (index >> 1) )->op[ index & 1 ]);
-	}
-	SynthHandler synthHandler;
-	Bit32u chanData;		//Frequency/octave and derived values
-	Bit32s old[2];			//Old data for feedback
-
-	Bit8u feedback;			//Feedback shift
-	Bit8u regB0;			//Register values to check for changes
-	Bit8u regC0;
-	//This should correspond with reg104, bit 6 indicates a Percussion channel, bit 7 indicates a silent channel
-	Bit8u fourMask;
-	Bit8s maskLeft;		//Sign extended values for both channel's panning
-	Bit8s maskRight;
-
-	//Forward the channel data to the operators of the channel
-	void SetChanData( const Chip* chip, Bit32u data );
-	//Change in the chandata, check for new values and if we have to forward to operators
-	void UpdateFrequency( const Chip* chip, Bit8u fourOp );
-	void WriteA0( const Chip* chip, Bit8u val );
-	void WriteB0( const Chip* chip, Bit8u val );
-	void WriteC0( const Chip* chip, Bit8u val );
-	void ResetC0( const Chip* chip );
-
-	//call this for the first channel
-	template< bool opl3Mode >
-	void GeneratePercussion( Chip* chip, Bit32s* output );
-
-	//Generate blocks of data in specific modes
-	template<SynthMode mode>
-	Channel* BlockTemplate( Chip* chip, Bit32u samples, Bit32s* output );
-	Channel();
-};
-
-struct Chip {
-	//This is used as the base counter for vibrato and tremolo
-	Bit32u lfoCounter;
-	Bit32u lfoAdd;
-
-
-	Bit32u noiseCounter;
-	Bit32u noiseAdd;
-	Bit32u noiseValue;
-
-	//Frequency scales for the different multiplications
-	Bit32u freqMul[16];
-	//Rates for decay and release for rate of this chip
-	Bit32u linearRates[76];
-	//Best match attack rates for the rate of this chip
-	Bit32u attackRates[76];
-
-	//18 channels with 2 operators each
-	Channel chan[18];
-
-	Bit8u reg104;
-	Bit8u reg08;
-	Bit8u reg04;
-	Bit8u regBD;
-	Bit8u vibratoIndex;
-	Bit8u tremoloIndex;
-	Bit8s vibratoSign;
-	Bit8u vibratoShift;
-	Bit8u tremoloValue;
-	Bit8u vibratoStrength;
-	Bit8u tremoloStrength;
-	//Mask for allowed wave forms
-	Bit8u waveFormMask;
-	//0 or -1 when enabled
-	Bit8s opl3Active;
-
-	//Return the maximum amount of samples before and LFO change
-	Bit32u ForwardLFO( Bit32u samples );
-	Bit32u ForwardNoise();
-
-	void WriteBD( Bit8u val );
-	void WriteReg(Bit32u reg, Bit8u val );
-
-	Bit32u WriteAddr( Bit32u port, Bit8u val );
-
-	void GenerateBlock2( Bitu samples, Bit32s* output );
-	void GenerateBlock3( Bitu samples, Bit32s* output );
-
-	void Generate( Bit32u samples );
-	void Setup( Bit32u r );
-
-	Chip();
-};
-
-void InitTables();
-
-}		//Namespace
-} // End of namespace DOSBox
-} // End of namespace OPL
-
-#endif // !DISABLE_DOSBOX_OPL
-
-#endif
diff --git a/sound/softsynth/opl/dosbox.cpp b/sound/softsynth/opl/dosbox.cpp
deleted file mode 100644
index 29993ce3d8..0000000000
--- a/sound/softsynth/opl/dosbox.cpp
+++ /dev/null
@@ -1,335 +0,0 @@
-/* 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.
- *
- * $URL$
- * $Id$
- */
-
-/*
- * Based on AdLib emulation code of DOSBox
- * Copyright (C) 2002-2009  The DOSBox Team
- * Licensed under GPLv2+
- * http://www.dosbox.com
- */
-
-#ifndef DISABLE_DOSBOX_OPL
-
-#include "dosbox.h"
-#include "dbopl.h"
-
-#include "common/system.h"
-#include "common/scummsys.h"
-
-#include <math.h>
-#include <string.h>
-
-namespace OPL {
-namespace DOSBox {
-
-Timer::Timer() {
-	masked = false;
-	overflow = false;
-	enabled = false;
-	counter = 0;
-	delay = 0;
-}
-
-void Timer::update(double time) {
-	if (!enabled || !delay)
-		return;
-	double deltaStart = time - startTime;
-	// Only set the overflow flag when not masked
-	if (deltaStart >= 0 && !masked)
-		overflow = 1;
-}
-
-void Timer::reset(double time) {
-	overflow = false;
-	if (!delay || !enabled)
-		return;
-	double delta = (time - startTime);
-	double rem = fmod(delta, delay);
-	double next = delay - rem;
-	startTime = time + next;
-}
-
-void Timer::stop() {
-	enabled = false;
-}
-
-void Timer::start(double time, int scale) {
-	//Don't enable again
-	if (enabled)
-		return;
-	enabled = true;
-	delay = 0.001 * (256 - counter) * scale;
-	startTime = time + delay;
-}
-
-bool Chip::write(uint32 reg, uint8 val) {
-	switch (reg) {
-	case 0x02:
-		timer[0].counter = val;
-		return true;
-	case 0x03:
-		timer[1].counter = val;
-		return true;
-	case 0x04:
-		double time = g_system->getMillis() / 1000.0;
-
-		if (val & 0x80) {
-			timer[0].reset(time);
-			timer[1].reset(time);
-		} else {
-			timer[0].update(time);
-			timer[1].update(time);
-
-			if (val & 0x1)
-				timer[0].start(time, 80);
-			else
-				timer[0].stop();
-
-			timer[0].masked = (val & 0x40) > 0;
-
-			if (timer[0].masked)
-				timer[0].overflow = false;
-
-			if (val & 0x2)
-				timer[1].start(time, 320);
-			else
-				timer[1].stop();
-
-			timer[1].masked = (val & 0x20) > 0;
-
-			if (timer[1].masked)
-				timer[1].overflow = false;
-		}
-		return true;
-	}
-	return false;
-}
-
-uint8 Chip::read() {
-	double time = g_system->getMillis() / 1000.0;
-
-	timer[0].update(time);
-	timer[1].update(time);
-
-	uint8 ret = 0;
-	// Overflow won't be set if a channel is masked
-	if (timer[0].overflow) {
-		ret |= 0x40;
-		ret |= 0x80;
-	}
-	if (timer[1].overflow) {
-		ret |= 0x20;
-		ret |= 0x80;
-	}
-	return ret;
-}
-
-OPL::OPL(Config::OplType type) : _type(type), _rate(0), _emulator(0) {
-}
-
-OPL::~OPL() {
-	free();
-}
-
-void OPL::free() {
-	delete _emulator;
-	_emulator = 0;
-}
-
-bool OPL::init(int rate) {
-	free();
-
-	memset(&_reg, 0, sizeof(_reg));
-	memset(_chip, 0, sizeof(_chip));
-
-	_emulator = new DBOPL::Chip();
-	if (!_emulator)
-		return false;
-
-	DBOPL::InitTables();
-	_emulator->Setup(rate);
-
-	if (_type == Config::kDualOpl2) {
-		// Setup opl3 mode in the hander
-		_emulator->WriteReg(0x105, 1);
-	}
-
-	_rate = rate;
-	return true;
-}
-
-void OPL::reset() {
-	init(_rate);
-}
-
-void OPL::write(int port, int val) {
-	if (port&1) {
-		switch (_type) {
-		case Config::kOpl2:
-		case Config::kOpl3:
-			if (!_chip[0].write(_reg.normal, val))
-				_emulator->WriteReg(_reg.normal, val);
-			break;
-		case Config::kDualOpl2:
-			// Not a 0x??8 port, then write to a specific port
-			if (!(port & 0x8)) {
-				byte index = (port & 2) >> 1;
-				dualWrite(index, _reg.dual[index], val);
-			} else {
-				//Write to both ports
-				dualWrite(0, _reg.dual[0], val);
-				dualWrite(1, _reg.dual[1], val);
-			}
-			break;
-		}
-	} else {
-		// Ask the handler to write the address
-		// Make sure to clip them in the right range
-		switch (_type) {
-		case Config::kOpl2:
-			_reg.normal = _emulator->WriteAddr(port, val) & 0xff;
-			break;
-		case Config::kOpl3:
-			_reg.normal = _emulator->WriteAddr(port, val) & 0x1ff;
-			break;
-		case Config::kDualOpl2:
-			// Not a 0x?88 port, when write to a specific side
-			if (!(port & 0x8)) {
-				byte index = (port & 2) >> 1;
-				_reg.dual[index] = val & 0xff;
-			} else {
-				_reg.dual[0] = val & 0xff;
-				_reg.dual[1] = val & 0xff;
-			}
-			break;
-		}
-	}
-}
-
-byte OPL::read(int port) {
-	switch (_type) {
-	case Config::kOpl2:
-		if (!(port & 1))
-			//Make sure the low bits are 6 on opl2
-			return _chip[0].read() | 0x6;
-		break;
-	case Config::kOpl3:
-		if (!(port & 1))
-			return _chip[0].read();
-		break;
-	case Config::kDualOpl2:
-		// Only return for the lower ports
-		if (port & 1)
-			return 0xff;
-		// Make sure the low bits are 6 on opl2
-		return _chip[(port >> 1) & 1].read() | 0x6;
-	}
-	return 0;
-}
-
-void OPL::writeReg(int r, int v) {
-	byte tempReg = 0;
-	switch (_type) {
-	case Config::kOpl2:
-	case Config::kDualOpl2:
-	case Config::kOpl3:
-		// We can't use _handler->writeReg here directly, since it would miss timer changes.
-
-		// Backup old setup register
-		tempReg = _reg.normal;
-
-		// We need to set the register we want to write to via port 0x388
-		write(0x388, r);
-		// Do the real writing to the register
-		write(0x389, v);
-		// Restore the old register
-		write(0x388, tempReg);
-		break;
-	};
-}
-
-void OPL::dualWrite(uint8 index, uint8 reg, uint8 val) {
-	// Make sure you don't use opl3 features
-	// Don't allow write to disable opl3
-	if (reg == 5)
-		return;
-
-	// Only allow 4 waveforms
-	if (reg >= 0xE0 && reg <= 0xE8)
-		val &= 3;
-
-	// Write to the timer?
-	if (_chip[index].write(reg, val))
-		return;
-
-	// Enabling panning
-	if (reg >= 0xC0 && reg <= 0xC8) {
-		val &= 15;
-		val |= index ? 0xA0 : 0x50;
-	}
-
-	uint32 fullReg = reg + (index ? 0x100 : 0);
-	_emulator->WriteReg(fullReg, val);
-}
-
-void OPL::readBuffer(int16 *buffer, int length) {
-	// For stereo OPL cards, we divide the sample count by 2,
-	// to match stereo AudioStream behavior.
-	if (_type != Config::kOpl2)
-		length >>= 1;
-
-	const uint bufferLength = 512;
-	int32 tempBuffer[bufferLength * 2];
-
-	if (_emulator->opl3Active) {
-		while (length > 0) {
-			const uint readSamples = MIN<uint>(length, bufferLength);
-
-			_emulator->GenerateBlock3(readSamples, tempBuffer);
-
-			for (uint i = 0; i < (readSamples << 1); ++i)
-				buffer[i] = tempBuffer[i];
-
-			buffer += (readSamples << 1);
-			length -= readSamples;
-		}
-	} else {
-		while (length > 0) {
-			const uint readSamples = MIN<uint>(length, bufferLength << 1);
-
-			_emulator->GenerateBlock2(readSamples, tempBuffer);
-
-			for (uint i = 0; i < readSamples; ++i)
-				buffer[i] = tempBuffer[i];
-
-			buffer += readSamples;
-			length -= readSamples;
-		}
-	}
-}
-
-} // End of namespace DOSBox
-} // End of namespace OPL
-
-#endif // !DISABLE_DOSBOX_ADLIB
diff --git a/sound/softsynth/opl/dosbox.h b/sound/softsynth/opl/dosbox.h
deleted file mode 100644
index 958310611d..0000000000
--- a/sound/softsynth/opl/dosbox.h
+++ /dev/null
@@ -1,110 +0,0 @@
-/* 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.
- *
- * $URL$
- * $Id$
- */
-
-/*
- * Based on OPL emulation code of DOSBox
- * Copyright (C) 2002-2009  The DOSBox Team
- * Licensed under GPLv2+
- * http://www.dosbox.com
- */
-
-#ifndef SOUND_SOFTSYNTH_OPL_DOSBOX_H
-#define SOUND_SOFTSYNTH_OPL_DOSBOX_H
-
-#ifndef DISABLE_DOSBOX_OPL
-
-#include "sound/fmopl.h"
-
-namespace OPL {
-namespace DOSBox {
-
-struct Timer {
-	double startTime;
-	double delay;
-	bool enabled, overflow, masked;
-	uint8 counter;
-
-	Timer();
-
-	//Call update before making any further changes
-	void update(double time);
-
-	//On a reset make sure the start is in sync with the next cycle
-	void reset(double time);
-
-	void stop();
-
-	void start(double time, int scale);
-};
-
-struct Chip {
-	//Last selected register
-	Timer timer[2];
-	//Check for it being a write to the timer
-	bool write(uint32 addr, uint8 val);
-	//Read the current timer state, will use current double
-	uint8 read();
-};
-
-namespace DBOPL {
-struct Chip;
-} // end of namespace DBOPL
-
-class OPL : public ::OPL::OPL {
-private:
-	Config::OplType _type;
-	uint _rate;
-
-	DBOPL::Chip *_emulator;
-	Chip _chip[2];
-	union {
-		uint16 normal;
-		uint8 dual[2];
-	} _reg;
-
-	void free();
-	void dualWrite(uint8 index, uint8 reg, uint8 val);
-public:
-	OPL(Config::OplType type);
-	~OPL();
-
-	bool init(int rate);
-	void reset();
-
-	void write(int a, int v);
-	byte read(int a);
-
-	void writeReg(int r, int v);
-
-	void readBuffer(int16 *buffer, int length);
-	bool isStereo() const { return _type != Config::kOpl2; }
-};
-
-} // End of namespace DOSBox
-} // End of namespace OPL
-
-#endif // !DISABLE_DOSBOX_OPL
-
-#endif
-
diff --git a/sound/softsynth/opl/mame.cpp b/sound/softsynth/opl/mame.cpp
deleted file mode 100644
index c875080e8f..0000000000
--- a/sound/softsynth/opl/mame.cpp
+++ /dev/null
@@ -1,1234 +0,0 @@
-/* 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.
- *
- * $URL$
- * $Id$
- *
- * LGPL licensed version of MAMEs fmopl (V0.37a modified) by
- * Tatsuyuki Satoh. Included from LGPL'ed AdPlug.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <stdarg.h>
-#include <math.h>
-
-#include "mame.h"
-
-#if defined (_WIN32_WCE) || defined (__SYMBIAN32__) || defined(__GP32__) || defined(GP2X) || defined (__MAEMO__) || defined(__DS__) || defined (__MINT__) || defined(__N64__)
-#include "common/config-manager.h"
-#endif
-
-#if defined(__DS__)
-#include "dsmain.h"
-#endif
-
-namespace OPL {
-namespace MAME {
-
-OPL::~OPL() {
-	MAME::OPLDestroy(_opl);
-	_opl = 0;
-}
-
-bool OPL::init(int rate) {
-	if (_opl)
-		MAME::OPLDestroy(_opl);
-
-	_opl = MAME::makeAdLibOPL(rate);
-	return (_opl != 0);
-}
-
-void OPL::reset() {
-	MAME::OPLResetChip(_opl);
-}
-
-void OPL::write(int a, int v) {
-	MAME::OPLWrite(_opl, a, v);
-}
-
-byte OPL::read(int a) {
-	return MAME::OPLRead(_opl, a);
-}
-
-void OPL::writeReg(int r, int v) {
-	MAME::OPLWriteReg(_opl, r, v);
-}
-
-void OPL::readBuffer(int16 *buffer, int length) {
-	MAME::YM3812UpdateOne(_opl, buffer, length);
-}
-
-/* -------------------- preliminary define section --------------------- */
-/* attack/decay rate time rate */
-#define OPL_ARRATE     141280  /* RATE 4 =  2826.24ms @ 3.6MHz */
-#define OPL_DRRATE    1956000  /* RATE 4 = 39280.64ms @ 3.6MHz */
-
-#define FREQ_BITS 24			/* frequency turn          */
-
-/* counter bits = 20 , octerve 7 */
-#define FREQ_RATE   (1<<(FREQ_BITS-20))
-#define TL_BITS    (FREQ_BITS+2)
-
-/* final output shift , limit minimum and maximum */
-#define OPL_OUTSB   (TL_BITS+3-16)		/* OPL output final shift 16bit */
-#define OPL_MAXOUT (0x7fff<<OPL_OUTSB)
-#define OPL_MINOUT (-0x8000<<OPL_OUTSB)
-
-/* -------------------- quality selection --------------------- */
-
-/* sinwave entries */
-/* used static memory = SIN_ENT * 4 (byte) */
-#ifdef __DS__
-#define SIN_ENT_SHIFT 8
-#else
-#define SIN_ENT_SHIFT 11
-#endif
-#define SIN_ENT (1<<SIN_ENT_SHIFT)
-
-/* output level entries (envelope,sinwave) */
-/* envelope counter lower bits */
-int ENV_BITS;
-/* envelope output entries */
-int EG_ENT;
-
-/* used dynamic memory = EG_ENT*4*4(byte)or EG_ENT*6*4(byte) */
-/* used static  memory = EG_ENT*4 (byte)                     */
-int EG_OFF;								 /* OFF */
-int EG_DED;
-int EG_DST;								 /* DECAY START */
-int EG_AED;
-#define EG_AST   0                       /* ATTACK START */
-
-#define EG_STEP (96.0/EG_ENT) /* OPL is 0.1875 dB step  */
-
-/* LFO table entries */
-#define VIB_ENT 512
-#define VIB_SHIFT (32-9)
-#define AMS_ENT 512
-#define AMS_SHIFT (32-9)
-
-#define VIB_RATE_SHIFT 8
-#define VIB_RATE (1<<VIB_RATE_SHIFT)
-
-/* -------------------- local defines , macros --------------------- */
-
-/* register number to channel number , slot offset */
-#define SLOT1 0
-#define SLOT2 1
-
-/* envelope phase */
-#define ENV_MOD_RR  0x00
-#define ENV_MOD_DR  0x01
-#define ENV_MOD_AR  0x02
-
-/* -------------------- tables --------------------- */
-static const int slot_array[32] = {
-	 0, 2, 4, 1, 3, 5,-1,-1,
-	 6, 8,10, 7, 9,11,-1,-1,
-	12,14,16,13,15,17,-1,-1,
-	-1,-1,-1,-1,-1,-1,-1,-1
-};
-
-static uint KSL_TABLE[8 * 16];
-
-static const double KSL_TABLE_SEED[8 * 16] = {
-	/* OCT 0 */
-	0.000, 0.000, 0.000, 0.000,
-	0.000, 0.000, 0.000, 0.000,
-	0.000, 0.000, 0.000, 0.000,
-	0.000, 0.000, 0.000, 0.000,
-	/* OCT 1 */
-	0.000, 0.000, 0.000, 0.000,
-	0.000, 0.000, 0.000, 0.000,
-	0.000, 0.750, 1.125, 1.500,
-	1.875, 2.250, 2.625, 3.000,
-	/* OCT 2 */
-	0.000, 0.000, 0.000, 0.000,
-	0.000, 1.125, 1.875, 2.625,
-	3.000, 3.750, 4.125, 4.500,
-	4.875, 5.250, 5.625, 6.000,
-	/* OCT 3 */
-	0.000, 0.000, 0.000, 1.875,
-	3.000, 4.125, 4.875, 5.625,
-	6.000, 6.750, 7.125, 7.500,
-	7.875, 8.250, 8.625, 9.000,
-	/* OCT 4 */
-	0.000, 0.000, 3.000, 4.875,
-	6.000, 7.125, 7.875, 8.625,
-	9.000, 9.750, 10.125, 10.500,
-	10.875, 11.250, 11.625, 12.000,
-	/* OCT 5 */
-	0.000, 3.000, 6.000, 7.875,
-	9.000, 10.125, 10.875, 11.625,
-	12.000, 12.750, 13.125, 13.500,
-	13.875, 14.250, 14.625, 15.000,
-	/* OCT 6 */
-	0.000, 6.000, 9.000, 10.875,
-	12.000, 13.125, 13.875, 14.625,
-	15.000, 15.750, 16.125, 16.500,
-	16.875, 17.250, 17.625, 18.000,
-	/* OCT 7 */
-	0.000, 9.000, 12.000, 13.875,
-	15.000, 16.125, 16.875, 17.625,
-	18.000, 18.750, 19.125, 19.500,
-	19.875, 20.250, 20.625, 21.000
-};
-
-/* sustain level table (3db per step) */
-/* 0 - 15: 0, 3, 6, 9,12,15,18,21,24,27,30,33,36,39,42,93 (dB)*/
-
-static int SL_TABLE[16];
-
-static const uint SL_TABLE_SEED[16] = {
-	0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 31
-};
-
-#define TL_MAX (EG_ENT * 2) /* limit(tl + ksr + envelope) + sinwave */
-/* TotalLevel : 48 24 12  6  3 1.5 0.75 (dB) */
-/* TL_TABLE[ 0      to TL_MAX          ] : plus  section */
-/* TL_TABLE[ TL_MAX to TL_MAX+TL_MAX-1 ] : minus section */
-static int *TL_TABLE;
-
-/* pointers to TL_TABLE with sinwave output offset */
-static int **SIN_TABLE;
-
-/* LFO table */
-static int *AMS_TABLE;
-static int *VIB_TABLE;
-
-/* envelope output curve table */
-/* attack + decay + OFF */
-//static int ENV_CURVE[2*EG_ENT+1];
-//static int ENV_CURVE[2 * 4096 + 1];   // to keep it static ...
-static int *ENV_CURVE;
-
-
-/* multiple table */
-#define ML(a) (int)(a * 2)
-static const uint MUL_TABLE[16]= {
-/* 1/2, 1, 2, 3, 4, 5, 6, 7, 8, 9,10,11,12,13,14,15 */
-	ML(0.50), ML(1.00), ML(2.00),  ML(3.00), ML(4.00), ML(5.00), ML(6.00), ML(7.00),
-	ML(8.00), ML(9.00), ML(10.00), ML(10.00),ML(12.00),ML(12.00),ML(15.00),ML(15.00)
-};
-#undef ML
-
-/* dummy attack / decay rate ( when rate == 0 ) */
-static int RATE_0[16]=
-{0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0};
-
-/* -------------------- static state --------------------- */
-
-/* lock level of common table */
-static int num_lock = 0;
-
-/* work table */
-static void *cur_chip = NULL;	/* current chip point */
-/* currenct chip state */
-/* static OPLSAMPLE  *bufL,*bufR; */
-static OPL_CH *S_CH;
-static OPL_CH *E_CH;
-OPL_SLOT *SLOT7_1, *SLOT7_2, *SLOT8_1, *SLOT8_2;
-
-static int outd[1];
-static int ams;
-static int vib;
-int *ams_table;
-int *vib_table;
-static int amsIncr;
-static int vibIncr;
-static int feedback2;		/* connect for SLOT 2 */
-
-/* --------------------- rebuild tables ------------------- */
-
-#define SC_KSL(mydb) ((uint) (mydb / (EG_STEP / 2)))
-#define SC_SL(db) (int)(db * ((3 / EG_STEP) * (1 << ENV_BITS))) + EG_DST
-
-void OPLBuildTables(int ENV_BITS_PARAM, int EG_ENT_PARAM) {
-	int i;
-
-	ENV_BITS = ENV_BITS_PARAM;
-	EG_ENT = EG_ENT_PARAM;
-	EG_OFF = ((2 * EG_ENT)<<ENV_BITS);  /* OFF          */
-	EG_DED = EG_OFF;
-	EG_DST = (EG_ENT << ENV_BITS);     /* DECAY  START */
-	EG_AED = EG_DST;
-	//EG_STEP = (96.0/EG_ENT);
-
-	for (i = 0; i < ARRAYSIZE(KSL_TABLE_SEED); i++)
-		KSL_TABLE[i] = SC_KSL(KSL_TABLE_SEED[i]);
-
-	for (i = 0; i < ARRAYSIZE(SL_TABLE_SEED); i++)
-		SL_TABLE[i] = SC_SL(SL_TABLE_SEED[i]);
-}
-
-#undef SC_KSL
-#undef SC_SL
-
-/* --------------------- subroutines  --------------------- */
-
-/* status set and IRQ handling */
-inline void OPL_STATUS_SET(FM_OPL *OPL, int flag) {
-	/* set status flag */
-	OPL->status |= flag;
-	if (!(OPL->status & 0x80)) {
-		if (OPL->status & OPL->statusmask) {	/* IRQ on */
-			OPL->status |= 0x80;
-			/* callback user interrupt handler (IRQ is OFF to ON) */
-			if (OPL->IRQHandler)
-				(OPL->IRQHandler)(OPL->IRQParam,1);
-		}
-	}
-}
-
-/* status reset and IRQ handling */
-inline void OPL_STATUS_RESET(FM_OPL *OPL, int flag) {
-	/* reset status flag */
-	OPL->status &= ~flag;
-	if ((OPL->status & 0x80)) {
-		if (!(OPL->status & OPL->statusmask)) {
-			OPL->status &= 0x7f;
-			/* callback user interrupt handler (IRQ is ON to OFF) */
-			if (OPL->IRQHandler) (OPL->IRQHandler)(OPL->IRQParam,0);
-		}
-	}
-}
-
-/* IRQ mask set */
-inline void OPL_STATUSMASK_SET(FM_OPL *OPL, int flag) {
-	OPL->statusmask = flag;
-	/* IRQ handling check */
-	OPL_STATUS_SET(OPL,0);
-	OPL_STATUS_RESET(OPL,0);
-}
-
-/* ----- key on  ----- */
-inline void OPL_KEYON(OPL_SLOT *SLOT) {
-	/* sin wave restart */
-	SLOT->Cnt = 0;
-	/* set attack */
-	SLOT->evm = ENV_MOD_AR;
-	SLOT->evs = SLOT->evsa;
-	SLOT->evc = EG_AST;
-	SLOT->eve = EG_AED;
-}
-
-/* ----- key off ----- */
-inline void OPL_KEYOFF(OPL_SLOT *SLOT) {
-	if (SLOT->evm > ENV_MOD_RR) {
-		/* set envelope counter from envleope output */
-
-		// WORKAROUND: The Kyra engine does something very strange when
-		// starting a new song. For each channel:
-		//
-		// * The release rate is set to "fastest".
-		// * Any note is keyed off.
-		// * A very low-frequency note is keyed on.
-		//
-		// Usually, what happens next is that the real notes is keyed
-		// on immediately, in which case there's no problem.
-		//
-		// However, if the note is again keyed off (because the channel
-		// begins on a rest rather than a note), the envelope counter
-		// was moved from the very lowest point on the attack curve to
-		// the very highest point on the release curve.
-		//
-		// Again, this might not be a problem, if the release rate is
-		// still set to "fastest". But in many cases, it had already
-		// been increased. And, possibly because of inaccuracies in the
-		// envelope generator, that would cause the note to "fade out"
-		// for quite a long time.
-		//
-		// What we really need is a way to find the correct starting
-		// point for the envelope counter, and that may be what the
-		// commented-out line below is meant to do. For now, simply
-		// handle the pathological case.
-
-		if (SLOT->evm == ENV_MOD_AR && SLOT->evc == EG_AST)
-			SLOT->evc = EG_DED;
-		else if (!(SLOT->evc & EG_DST))
-			//SLOT->evc = (ENV_CURVE[SLOT->evc>>ENV_BITS]<<ENV_BITS) + EG_DST;
-			SLOT->evc = EG_DST;
-		SLOT->eve = EG_DED;
-		SLOT->evs = SLOT->evsr;
-		SLOT->evm = ENV_MOD_RR;
-	}
-}
-
-/* ---------- calcrate Envelope Generator & Phase Generator ---------- */
-
-/* return : envelope output */
-inline uint OPL_CALC_SLOT(OPL_SLOT *SLOT) {
-	/* calcrate envelope generator */
-	if ((SLOT->evc += SLOT->evs) >= SLOT->eve) {
-		switch (SLOT->evm) {
-		case ENV_MOD_AR: /* ATTACK -> DECAY1 */
-			/* next DR */
-			SLOT->evm = ENV_MOD_DR;
-			SLOT->evc = EG_DST;
-			SLOT->eve = SLOT->SL;
-			SLOT->evs = SLOT->evsd;
-			break;
-		case ENV_MOD_DR: /* DECAY -> SL or RR */
-			SLOT->evc = SLOT->SL;
-			SLOT->eve = EG_DED;
-			if (SLOT->eg_typ) {
-				SLOT->evs = 0;
-			} else {
-				SLOT->evm = ENV_MOD_RR;
-				SLOT->evs = SLOT->evsr;
-			}
-			break;
-		case ENV_MOD_RR: /* RR -> OFF */
-			SLOT->evc = EG_OFF;
-			SLOT->eve = EG_OFF + 1;
-			SLOT->evs = 0;
-			break;
-		}
-	}
-	/* calcrate envelope */
-	return SLOT->TLL + ENV_CURVE[SLOT->evc>>ENV_BITS] + (SLOT->ams ? ams : 0);
-}
-
-/* set algorythm connection */
-static void set_algorythm(OPL_CH *CH) {
-	int *carrier = &outd[0];
-	CH->connect1 = CH->CON ? carrier : &feedback2;
-	CH->connect2 = carrier;
-}
-
-/* ---------- frequency counter for operater update ---------- */
-inline void CALC_FCSLOT(OPL_CH *CH, OPL_SLOT *SLOT) {
-	int ksr;
-
-	/* frequency step counter */
-	SLOT->Incr = CH->fc * SLOT->mul;
-	ksr = CH->kcode >> SLOT->KSR;
-
-	if (SLOT->ksr != ksr) {
-		SLOT->ksr = ksr;
-		/* attack , decay rate recalcration */
-		SLOT->evsa = SLOT->AR[ksr];
-		SLOT->evsd = SLOT->DR[ksr];
-		SLOT->evsr = SLOT->RR[ksr];
-	}
-	SLOT->TLL = SLOT->TL + (CH->ksl_base>>SLOT->ksl);
-}
-
-/* set multi,am,vib,EG-TYP,KSR,mul */
-inline void set_mul(FM_OPL *OPL, int slot, int v) {
-	OPL_CH   *CH   = &OPL->P_CH[slot>>1];
-	OPL_SLOT *SLOT = &CH->SLOT[slot & 1];
-
-	SLOT->mul    = MUL_TABLE[v & 0x0f];
-	SLOT->KSR    = (v & 0x10) ? 0 : 2;
-	SLOT->eg_typ = (v & 0x20) >> 5;
-	SLOT->vib    = (v & 0x40);
-	SLOT->ams    = (v & 0x80);
-	CALC_FCSLOT(CH, SLOT);
-}
-
-/* set ksl & tl */
-inline void set_ksl_tl(FM_OPL *OPL, int slot, int v) {
-	OPL_CH   *CH   = &OPL->P_CH[slot>>1];
-	OPL_SLOT *SLOT = &CH->SLOT[slot & 1];
-	int ksl = v >> 6; /* 0 / 1.5 / 3 / 6 db/OCT */
-
-	SLOT->ksl = ksl ? 3-ksl : 31;
-	SLOT->TL  = (int)((v & 0x3f) * (0.75 / EG_STEP)); /* 0.75db step */
-
-	if (!(OPL->mode & 0x80)) {	/* not CSM latch total level */
-		SLOT->TLL = SLOT->TL + (CH->ksl_base >> SLOT->ksl);
-	}
-}
-
-/* set attack rate & decay rate  */
-inline void set_ar_dr(FM_OPL *OPL, int slot, int v) {
-	OPL_CH   *CH   = &OPL->P_CH[slot>>1];
-	OPL_SLOT *SLOT = &CH->SLOT[slot & 1];
-	int ar = v >> 4;
-	int dr = v & 0x0f;
-
-	SLOT->AR = ar ? &OPL->AR_TABLE[ar << 2] : RATE_0;
-	SLOT->evsa = SLOT->AR[SLOT->ksr];
-	if (SLOT->evm == ENV_MOD_AR)
-		SLOT->evs = SLOT->evsa;
-
-	SLOT->DR = dr ? &OPL->DR_TABLE[dr<<2] : RATE_0;
-	SLOT->evsd = SLOT->DR[SLOT->ksr];
-	if (SLOT->evm == ENV_MOD_DR)
-		SLOT->evs = SLOT->evsd;
-}
-
-/* set sustain level & release rate */
-inline void set_sl_rr(FM_OPL *OPL, int slot, int v) {
-	OPL_CH   *CH   = &OPL->P_CH[slot>>1];
-	OPL_SLOT *SLOT = &CH->SLOT[slot & 1];
-	int sl = v >> 4;
-	int rr = v & 0x0f;
-
-	SLOT->SL = SL_TABLE[sl];
-	if (SLOT->evm == ENV_MOD_DR)
-		SLOT->eve = SLOT->SL;
-	SLOT->RR = &OPL->DR_TABLE[rr<<2];
-	SLOT->evsr = SLOT->RR[SLOT->ksr];
-	if (SLOT->evm == ENV_MOD_RR)
-		SLOT->evs = SLOT->evsr;
-}
-
-/* operator output calcrator */
-
-#define OP_OUT(slot,env,con)   slot->wavetable[((slot->Cnt + con)>>(24-SIN_ENT_SHIFT)) & (SIN_ENT-1)][env]
-/* ---------- calcrate one of channel ---------- */
-inline void OPL_CALC_CH(OPL_CH *CH) {
-	uint env_out;
-	OPL_SLOT *SLOT;
-
-	feedback2 = 0;
-	/* SLOT 1 */
-	SLOT = &CH->SLOT[SLOT1];
-	env_out=OPL_CALC_SLOT(SLOT);
-	if (env_out < (uint)(EG_ENT - 1)) {
-		/* PG */
-		if (SLOT->vib)
-			SLOT->Cnt += (SLOT->Incr * vib) >> VIB_RATE_SHIFT;
-		else
-			SLOT->Cnt += SLOT->Incr;
-		/* connection */
-		if (CH->FB) {
-			int feedback1 = (CH->op1_out[0] + CH->op1_out[1]) >> CH->FB;
-			CH->op1_out[1] = CH->op1_out[0];
-			*CH->connect1 += CH->op1_out[0] = OP_OUT(SLOT, env_out, feedback1);
-		} else {
-			*CH->connect1 += OP_OUT(SLOT, env_out, 0);
-		}
-	} else {
-		CH->op1_out[1] = CH->op1_out[0];
-		CH->op1_out[0] = 0;
-	}
-	/* SLOT 2 */
-	SLOT = &CH->SLOT[SLOT2];
-	env_out=OPL_CALC_SLOT(SLOT);
-	if (env_out < (uint)(EG_ENT - 1)) {
-		/* PG */
-		if (SLOT->vib)
-			SLOT->Cnt += (SLOT->Incr * vib) >> VIB_RATE_SHIFT;
-		else
-			SLOT->Cnt += SLOT->Incr;
-		/* connection */
-		outd[0] += OP_OUT(SLOT, env_out, feedback2);
-	}
-}
-
-/* ---------- calcrate rythm block ---------- */
-#define WHITE_NOISE_db 6.0
-inline void OPL_CALC_RH(FM_OPL *OPL, OPL_CH *CH) {
-	uint env_tam, env_sd, env_top, env_hh;
-	// This code used to do int(OPL->rnd.getRandomBit() * (WHITE_NOISE_db / EG_STEP)),
-	// but EG_STEP = 96.0/EG_ENT, and WHITE_NOISE_db=6.0. So, that's equivalent to
-	// int(OPL->rnd.getRandomBit() * EG_ENT/16). We know that EG_ENT is 4096, or 1024,
-	// or 128, so we can safely avoid any FP ops.
-	int whitenoise = OPL->rnd.getRandomBit() * (EG_ENT>>4);
-
-	int tone8;
-
-	OPL_SLOT *SLOT;
-	int env_out;
-
-	/* BD : same as FM serial mode and output level is large */
-	feedback2 = 0;
-	/* SLOT 1 */
-	SLOT = &CH[6].SLOT[SLOT1];
-	env_out = OPL_CALC_SLOT(SLOT);
-	if (env_out < EG_ENT-1) {
-		/* PG */
-		if (SLOT->vib)
-			SLOT->Cnt += (SLOT->Incr * vib) >> VIB_RATE_SHIFT;
-		else
-			SLOT->Cnt += SLOT->Incr;
-		/* connection */
-		if (CH[6].FB) {
-			int feedback1 = (CH[6].op1_out[0] + CH[6].op1_out[1]) >> CH[6].FB;
-			CH[6].op1_out[1] = CH[6].op1_out[0];
-			feedback2 = CH[6].op1_out[0] = OP_OUT(SLOT, env_out, feedback1);
-		}
-		else {
-			feedback2 = OP_OUT(SLOT, env_out, 0);
-		}
-	} else {
-		feedback2 = 0;
-		CH[6].op1_out[1] = CH[6].op1_out[0];
-		CH[6].op1_out[0] = 0;
-	}
-	/* SLOT 2 */
-	SLOT = &CH[6].SLOT[SLOT2];
-	env_out = OPL_CALC_SLOT(SLOT);
-	if (env_out < EG_ENT-1) {
-		/* PG */
-		if (SLOT->vib)
-			SLOT->Cnt += (SLOT->Incr * vib) >> VIB_RATE_SHIFT;
-		else
-			SLOT->Cnt += SLOT->Incr;
-		/* connection */
-		outd[0] += OP_OUT(SLOT, env_out, feedback2) * 2;
-	}
-
-	// SD  (17) = mul14[fnum7] + white noise
-	// TAM (15) = mul15[fnum8]
-	// TOP (18) = fnum6(mul18[fnum8]+whitenoise)
-	// HH  (14) = fnum7(mul18[fnum8]+whitenoise) + white noise
-	env_sd = OPL_CALC_SLOT(SLOT7_2) + whitenoise;
-	env_tam =OPL_CALC_SLOT(SLOT8_1);
-	env_top = OPL_CALC_SLOT(SLOT8_2);
-	env_hh = OPL_CALC_SLOT(SLOT7_1) + whitenoise;
-
-	/* PG */
-	if (SLOT7_1->vib)
-		SLOT7_1->Cnt += (SLOT7_1->Incr * vib) >> (VIB_RATE_SHIFT-1);
-	else
-		SLOT7_1->Cnt += 2 * SLOT7_1->Incr;
-	if (SLOT7_2->vib)
-		SLOT7_2->Cnt += (CH[7].fc * vib) >> (VIB_RATE_SHIFT-3);
-	else
-		SLOT7_2->Cnt += (CH[7].fc * 8);
-	if (SLOT8_1->vib)
-		SLOT8_1->Cnt += (SLOT8_1->Incr * vib) >> VIB_RATE_SHIFT;
-	else
-		SLOT8_1->Cnt += SLOT8_1->Incr;
-	if (SLOT8_2->vib)
-		SLOT8_2->Cnt += ((CH[8].fc * 3) * vib) >> (VIB_RATE_SHIFT-4);
-	else
-		SLOT8_2->Cnt += (CH[8].fc * 48);
-
-	tone8 = OP_OUT(SLOT8_2,whitenoise,0 );
-
-	/* SD */
-	if (env_sd < (uint)(EG_ENT - 1))
-		outd[0] += OP_OUT(SLOT7_1, env_sd, 0) * 8;
-	/* TAM */
-	if (env_tam < (uint)(EG_ENT - 1))
-		outd[0] += OP_OUT(SLOT8_1, env_tam, 0) * 2;
-	/* TOP-CY */
-	if (env_top < (uint)(EG_ENT - 1))
-		outd[0] += OP_OUT(SLOT7_2, env_top, tone8) * 2;
-	/* HH */
-	if (env_hh  < (uint)(EG_ENT-1))
-		outd[0] += OP_OUT(SLOT7_2, env_hh, tone8) * 2;
-}
-
-/* ----------- initialize time tabls ----------- */
-static void init_timetables(FM_OPL *OPL, int ARRATE, int DRRATE) {
-	int i;
-	double rate;
-
-	/* make attack rate & decay rate tables */
-	for (i = 0; i < 4; i++)
-		OPL->AR_TABLE[i] = OPL->DR_TABLE[i] = 0;
-	for (i = 4; i <= 60; i++) {
-		rate = OPL->freqbase;						/* frequency rate */
-		if (i < 60)
-			rate *= 1.0 + (i & 3) * 0.25;		/* b0-1 : x1 , x1.25 , x1.5 , x1.75 */
-		rate *= 1 << ((i >> 2) - 1);						/* b2-5 : shift bit */
-		rate *= (double)(EG_ENT << ENV_BITS);
-		OPL->AR_TABLE[i] = (int)(rate / ARRATE);
-		OPL->DR_TABLE[i] = (int)(rate / DRRATE);
-	}
-	for (i = 60; i < 76; i++) {
-		OPL->AR_TABLE[i] = EG_AED-1;
-		OPL->DR_TABLE[i] = OPL->DR_TABLE[60];
-	}
-}
-
-/* ---------- generic table initialize ---------- */
-static int OPLOpenTable(void) {
-	int s,t;
-	double rate;
-	int i,j;
-	double pom;
-
-#ifdef __DS__
-	DS::fastRamReset();
-
-	TL_TABLE = (int *) DS::fastRamAlloc(TL_MAX * 2 * sizeof(int *));
-	SIN_TABLE = (int **) DS::fastRamAlloc(SIN_ENT * 4 * sizeof(int *));
-#else
-
-	/* allocate dynamic tables */
-	if ((TL_TABLE = (int *)malloc(TL_MAX * 2 * sizeof(int))) == NULL)
-		return 0;
-
-	if ((SIN_TABLE = (int **)malloc(SIN_ENT * 4 * sizeof(int *))) == NULL) {
-		free(TL_TABLE);
-		return 0;
-	}
-#endif
-
-	if ((AMS_TABLE = (int *)malloc(AMS_ENT * 2 * sizeof(int))) == NULL) {
-		free(TL_TABLE);
-		free(SIN_TABLE);
-		return 0;
-	}
-
-	if ((VIB_TABLE = (int *)malloc(VIB_ENT * 2 * sizeof(int))) == NULL) {
-		free(TL_TABLE);
-		free(SIN_TABLE);
-		free(AMS_TABLE);
-		return 0;
-	}
-	/* make total level table */
-	for (t = 0; t < EG_ENT - 1; t++) {
-		rate = ((1 << TL_BITS) - 1) / pow(10.0, EG_STEP * t / 20);	/* dB -> voltage */
-		TL_TABLE[         t] =  (int)rate;
-		TL_TABLE[TL_MAX + t] = -TL_TABLE[t];
-	}
-	/* fill volume off area */
-	for (t = EG_ENT - 1; t < TL_MAX; t++) {
-		TL_TABLE[t] = TL_TABLE[TL_MAX + t] = 0;
-	}
-
-	/* make sinwave table (total level offet) */
-	/* degree 0 = degree 180                   = off */
-	SIN_TABLE[0] = SIN_TABLE[SIN_ENT /2 ] = &TL_TABLE[EG_ENT - 1];
-	for (s = 1;s <= SIN_ENT / 4; s++) {
-		pom = sin(2 * PI * s / SIN_ENT); /* sin     */
-		pom = 20 * log10(1 / pom);	   /* decibel */
-		j = int(pom / EG_STEP);         /* TL_TABLE steps */
-
-		/* degree 0   -  90    , degree 180 -  90 : plus section */
-		SIN_TABLE[          s] = SIN_TABLE[SIN_ENT / 2 - s] = &TL_TABLE[j];
-		/* degree 180 - 270    , degree 360 - 270 : minus section */
-		SIN_TABLE[SIN_ENT / 2 + s] = SIN_TABLE[SIN_ENT - s] = &TL_TABLE[TL_MAX + j];
-	}
-	for (s = 0;s < SIN_ENT; s++) {
-		SIN_TABLE[SIN_ENT * 1 + s] = s < (SIN_ENT / 2) ? SIN_TABLE[s] : &TL_TABLE[EG_ENT];
-		SIN_TABLE[SIN_ENT * 2 + s] = SIN_TABLE[s % (SIN_ENT / 2)];
-		SIN_TABLE[SIN_ENT * 3 + s] = (s / (SIN_ENT / 4)) & 1 ? &TL_TABLE[EG_ENT] : SIN_TABLE[SIN_ENT * 2 + s];
-	}
-
-
-	ENV_CURVE = (int *)malloc(sizeof(int) * (2*EG_ENT+1));
-
-	/* envelope counter -> envelope output table */
-	for (i=0; i < EG_ENT; i++) {
-		/* ATTACK curve */
-		pom = pow(((double)(EG_ENT - 1 - i) / EG_ENT), 8) * EG_ENT;
-		/* if (pom >= EG_ENT) pom = EG_ENT-1; */
-		ENV_CURVE[i] = (int)pom;
-		/* DECAY ,RELEASE curve */
-		ENV_CURVE[(EG_DST >> ENV_BITS) + i]= i;
-	}
-	/* off */
-	ENV_CURVE[EG_OFF >> ENV_BITS]= EG_ENT - 1;
-	/* make LFO ams table */
-	for (i=0; i < AMS_ENT; i++) {
-		pom = (1.0 + sin(2 * PI * i / AMS_ENT)) / 2; /* sin */
-		AMS_TABLE[i]         = (int)((1.0 / EG_STEP) * pom); /* 1dB   */
-		AMS_TABLE[AMS_ENT + i] = (int)((4.8 / EG_STEP) * pom); /* 4.8dB */
-	}
-	/* make LFO vibrate table */
-	for (i=0; i < VIB_ENT; i++) {
-		/* 100cent = 1seminote = 6% ?? */
-		pom = (double)VIB_RATE * 0.06 * sin(2 * PI * i / VIB_ENT); /* +-100sect step */
-		VIB_TABLE[i]         = (int)(VIB_RATE + (pom * 0.07)); /* +- 7cent */
-		VIB_TABLE[VIB_ENT + i] = (int)(VIB_RATE + (pom * 0.14)); /* +-14cent */
-	}
-	return 1;
-}
-
-static void OPLCloseTable(void) {
-	free(TL_TABLE);
-	free(SIN_TABLE);
-	free(AMS_TABLE);
-	free(VIB_TABLE);
-	free(ENV_CURVE);
-}
-
-/* CSM Key Controll */
-inline void CSMKeyControll(OPL_CH *CH) {
-	OPL_SLOT *slot1 = &CH->SLOT[SLOT1];
-	OPL_SLOT *slot2 = &CH->SLOT[SLOT2];
-	/* all key off */
-	OPL_KEYOFF(slot1);
-	OPL_KEYOFF(slot2);
-	/* total level latch */
-	slot1->TLL = slot1->TL + (CH->ksl_base>>slot1->ksl);
-	slot1->TLL = slot1->TL + (CH->ksl_base>>slot1->ksl);
-	/* key on */
-	CH->op1_out[0] = CH->op1_out[1] = 0;
-	OPL_KEYON(slot1);
-	OPL_KEYON(slot2);
-}
-
-/* ---------- opl initialize ---------- */
-static void OPL_initalize(FM_OPL *OPL) {
-	int fn;
-
-	/* frequency base */
-	OPL->freqbase = (OPL->rate) ? ((double)OPL->clock / OPL->rate) / 72 : 0;
-	/* Timer base time */
-	OPL->TimerBase = 1.0/((double)OPL->clock / 72.0 );
-	/* make time tables */
-	init_timetables(OPL, OPL_ARRATE, OPL_DRRATE);
-	/* make fnumber -> increment counter table */
-	for (fn=0; fn < 1024; fn++) {
-		OPL->FN_TABLE[fn] = (uint)(OPL->freqbase * fn * FREQ_RATE * (1<<7) / 2);
-	}
-	/* LFO freq.table */
-	OPL->amsIncr = (int)(OPL->rate ? (double)AMS_ENT * (1 << AMS_SHIFT) / OPL->rate * 3.7 * ((double)OPL->clock/3600000) : 0);
-	OPL->vibIncr = (int)(OPL->rate ? (double)VIB_ENT * (1 << VIB_SHIFT) / OPL->rate * 6.4 * ((double)OPL->clock/3600000) : 0);
-}
-
-/* ---------- write a OPL registers ---------- */
-void OPLWriteReg(FM_OPL *OPL, int r, int v) {
-	OPL_CH *CH;
-	int slot;
-	uint block_fnum;
-
-	switch (r & 0xe0) {
-	case 0x00: /* 00-1f:controll */
-		switch (r & 0x1f) {
-		case 0x01:
-			/* wave selector enable */
-			if (OPL->type&OPL_TYPE_WAVESEL) {
-				OPL->wavesel = v & 0x20;
-				if (!OPL->wavesel) {
-					/* preset compatible mode */
-					int c;
-					for (c = 0; c < OPL->max_ch; c++) {
-						OPL->P_CH[c].SLOT[SLOT1].wavetable = &SIN_TABLE[0];
-						OPL->P_CH[c].SLOT[SLOT2].wavetable = &SIN_TABLE[0];
-					}
-				}
-			}
-			return;
-		case 0x02:	/* Timer 1 */
-			OPL->T[0] = (256-v) * 4;
-			break;
-		case 0x03:	/* Timer 2 */
-			OPL->T[1] = (256-v) * 16;
-			return;
-		case 0x04:	/* IRQ clear / mask and Timer enable */
-			if (v & 0x80) {	/* IRQ flag clear */
-				OPL_STATUS_RESET(OPL, 0x7f);
-			} else {	/* set IRQ mask ,timer enable*/
-				uint8 st1 = v & 1;
-				uint8 st2 = (v >> 1) & 1;
-				/* IRQRST,T1MSK,t2MSK,EOSMSK,BRMSK,x,ST2,ST1 */
-				OPL_STATUS_RESET(OPL, v & 0x78);
-				OPL_STATUSMASK_SET(OPL,((~v) & 0x78) | 0x01);
-				/* timer 2 */
-				if (OPL->st[1] != st2) {
-					double interval = st2 ? (double)OPL->T[1] * OPL->TimerBase : 0.0;
-					OPL->st[1] = st2;
-					if (OPL->TimerHandler) (OPL->TimerHandler)(OPL->TimerParam + 1, interval);
-				}
-				/* timer 1 */
-				if (OPL->st[0] != st1) {
-					double interval = st1 ? (double)OPL->T[0] * OPL->TimerBase : 0.0;
-					OPL->st[0] = st1;
-					if (OPL->TimerHandler) (OPL->TimerHandler)(OPL->TimerParam + 0, interval);
-				}
-			}
-			return;
-		}
-		break;
-	case 0x20:	/* am,vib,ksr,eg type,mul */
-		slot = slot_array[r&0x1f];
-		if (slot == -1)
-			return;
-		set_mul(OPL,slot,v);
-		return;
-	case 0x40:
-		slot = slot_array[r&0x1f];
-		if (slot == -1)
-			return;
-		set_ksl_tl(OPL,slot,v);
-		return;
-	case 0x60:
-		slot = slot_array[r&0x1f];
-		if (slot == -1)
-			return;
-		set_ar_dr(OPL,slot,v);
-		return;
-	case 0x80:
-		slot = slot_array[r&0x1f];
-		if (slot == -1)
-			return;
-		set_sl_rr(OPL,slot,v);
-		return;
-	case 0xa0:
-		switch (r) {
-		case 0xbd:
-			/* amsep,vibdep,r,bd,sd,tom,tc,hh */
-			{
-			uint8 rkey = OPL->rythm ^ v;
-			OPL->ams_table = &AMS_TABLE[v & 0x80 ? AMS_ENT : 0];
-			OPL->vib_table = &VIB_TABLE[v & 0x40 ? VIB_ENT : 0];
-			OPL->rythm  = v & 0x3f;
-			if (OPL->rythm & 0x20) {
-				/* BD key on/off */
-				if (rkey & 0x10) {
-					if (v & 0x10) {
-						OPL->P_CH[6].op1_out[0] = OPL->P_CH[6].op1_out[1] = 0;
-						OPL_KEYON(&OPL->P_CH[6].SLOT[SLOT1]);
-						OPL_KEYON(&OPL->P_CH[6].SLOT[SLOT2]);
-					} else {
-						OPL_KEYOFF(&OPL->P_CH[6].SLOT[SLOT1]);
-						OPL_KEYOFF(&OPL->P_CH[6].SLOT[SLOT2]);
-					}
-				}
-				/* SD key on/off */
-				if (rkey & 0x08) {
-					if (v & 0x08)
-						OPL_KEYON(&OPL->P_CH[7].SLOT[SLOT2]);
-					else
-						OPL_KEYOFF(&OPL->P_CH[7].SLOT[SLOT2]);
-				}/* TAM key on/off */
-				if (rkey & 0x04) {
-					if (v & 0x04)
-						OPL_KEYON(&OPL->P_CH[8].SLOT[SLOT1]);
-					else
-						OPL_KEYOFF(&OPL->P_CH[8].SLOT[SLOT1]);
-				}
-				/* TOP-CY key on/off */
-				if (rkey & 0x02) {
-					if (v & 0x02)
-						OPL_KEYON(&OPL->P_CH[8].SLOT[SLOT2]);
-					else
-						OPL_KEYOFF(&OPL->P_CH[8].SLOT[SLOT2]);
-				}
-				/* HH key on/off */
-				if (rkey & 0x01) {
-					if (v & 0x01)
-						OPL_KEYON(&OPL->P_CH[7].SLOT[SLOT1]);
-					else
-						OPL_KEYOFF(&OPL->P_CH[7].SLOT[SLOT1]);
-				}
-			}
-			}
-			return;
-
-		default:
-			break;
-		}
-		/* keyon,block,fnum */
-		if ((r & 0x0f) > 8)
-			return;
-		CH = &OPL->P_CH[r & 0x0f];
-		if (!(r&0x10)) {	/* a0-a8 */
-			block_fnum  = (CH->block_fnum & 0x1f00) | v;
-		} else {	/* b0-b8 */
-			int keyon = (v >> 5) & 1;
-			block_fnum = ((v & 0x1f) << 8) | (CH->block_fnum & 0xff);
-			if (CH->keyon != keyon) {
-				if ((CH->keyon=keyon)) {
-					CH->op1_out[0] = CH->op1_out[1] = 0;
-					OPL_KEYON(&CH->SLOT[SLOT1]);
-					OPL_KEYON(&CH->SLOT[SLOT2]);
-				} else {
-					OPL_KEYOFF(&CH->SLOT[SLOT1]);
-					OPL_KEYOFF(&CH->SLOT[SLOT2]);
-				}
-			}
-		}
-		/* update */
-		if (CH->block_fnum != block_fnum) {
-			int blockRv = 7 - (block_fnum >> 10);
-			int fnum = block_fnum & 0x3ff;
-			CH->block_fnum = block_fnum;
-			CH->ksl_base = KSL_TABLE[block_fnum >> 6];
-			CH->fc = OPL->FN_TABLE[fnum] >> blockRv;
-			CH->kcode = CH->block_fnum >> 9;
-			if ((OPL->mode & 0x40) && CH->block_fnum & 0x100)
-				CH->kcode |=1;
-			CALC_FCSLOT(CH,&CH->SLOT[SLOT1]);
-			CALC_FCSLOT(CH,&CH->SLOT[SLOT2]);
-		}
-		return;
-	case 0xc0:
-		/* FB,C */
-		if ((r & 0x0f) > 8)
-			return;
-		CH = &OPL->P_CH[r&0x0f];
-		{
-			int feedback = (v >> 1) & 7;
-			CH->FB = feedback ? (8 + 1) - feedback : 0;
-			CH->CON = v & 1;
-			set_algorythm(CH);
-		}
-		return;
-	case 0xe0: /* wave type */
-		slot = slot_array[r & 0x1f];
-		if (slot == -1)
-			return;
-		CH = &OPL->P_CH[slot>>1];
-		if (OPL->wavesel) {
-			CH->SLOT[slot&1].wavetable = &SIN_TABLE[(v & 0x03) * SIN_ENT];
-		}
-		return;
-	}
-}
-
-/* lock/unlock for common table */
-static int OPL_LockTable(void) {
-	num_lock++;
-	if (num_lock>1)
-		return 0;
-	/* first time */
-	cur_chip = NULL;
-	/* allocate total level table (128kb space) */
-	if (!OPLOpenTable()) {
-		num_lock--;
-		return -1;
-	}
-	return 0;
-}
-
-static void OPL_UnLockTable(void) {
-	if (num_lock)
-		num_lock--;
-	if (num_lock)
-		return;
-	/* last time */
-	cur_chip = NULL;
-	OPLCloseTable();
-}
-
-/*******************************************************************************/
-/*		YM3812 local section                                                   */
-/*******************************************************************************/
-
-/* ---------- update one of chip ----------- */
-void YM3812UpdateOne(FM_OPL *OPL, int16 *buffer, int length) {
-	int i;
-	int data;
-	int16 *buf = buffer;
-	uint amsCnt = OPL->amsCnt;
-	uint vibCnt = OPL->vibCnt;
-	uint8 rythm = OPL->rythm & 0x20;
-	OPL_CH *CH, *R_CH;
-
-
-	if ((void *)OPL != cur_chip) {
-		cur_chip = (void *)OPL;
-		/* channel pointers */
-		S_CH = OPL->P_CH;
-		E_CH = &S_CH[9];
-		/* rythm slot */
-		SLOT7_1 = &S_CH[7].SLOT[SLOT1];
-		SLOT7_2 = &S_CH[7].SLOT[SLOT2];
-		SLOT8_1 = &S_CH[8].SLOT[SLOT1];
-		SLOT8_2 = &S_CH[8].SLOT[SLOT2];
-		/* LFO state */
-		amsIncr = OPL->amsIncr;
-		vibIncr = OPL->vibIncr;
-		ams_table = OPL->ams_table;
-		vib_table = OPL->vib_table;
-	}
-	R_CH = rythm ? &S_CH[6] : E_CH;
-	for (i = 0; i < length; i++) {
-		/*            channel A         channel B         channel C      */
-		/* LFO */
-		ams = ams_table[(amsCnt += amsIncr) >> AMS_SHIFT];
-		vib = vib_table[(vibCnt += vibIncr) >> VIB_SHIFT];
-		outd[0] = 0;
-		/* FM part */
-		for (CH = S_CH; CH < R_CH; CH++)
-			OPL_CALC_CH(CH);
-		/* Rythn part */
-		if (rythm)
-			OPL_CALC_RH(OPL, S_CH);
-		/* limit check */
-		data = CLIP(outd[0], OPL_MINOUT, OPL_MAXOUT);
-		/* store to sound buffer */
-		buf[i] = data >> OPL_OUTSB;
-	}
-
-	OPL->amsCnt = amsCnt;
-	OPL->vibCnt = vibCnt;
-}
-
-/* ---------- reset a chip ---------- */
-void OPLResetChip(FM_OPL *OPL) {
-	int c,s;
-	int i;
-
-	/* reset chip */
-	OPL->mode = 0;	/* normal mode */
-	OPL_STATUS_RESET(OPL, 0x7f);
-	/* reset with register write */
-	OPLWriteReg(OPL, 0x01,0); /* wabesel disable */
-	OPLWriteReg(OPL, 0x02,0); /* Timer1 */
-	OPLWriteReg(OPL, 0x03,0); /* Timer2 */
-	OPLWriteReg(OPL, 0x04,0); /* IRQ mask clear */
-	for (i = 0xff; i >= 0x20; i--)
-		OPLWriteReg(OPL,i,0);
-	/* reset OPerator parameter */
-	for (c = 0; c < OPL->max_ch; c++) {
-		OPL_CH *CH = &OPL->P_CH[c];
-		/* OPL->P_CH[c].PAN = OPN_CENTER; */
-		for (s = 0; s < 2; s++) {
-			/* wave table */
-			CH->SLOT[s].wavetable = &SIN_TABLE[0];
-			/* CH->SLOT[s].evm = ENV_MOD_RR; */
-			CH->SLOT[s].evc = EG_OFF;
-			CH->SLOT[s].eve = EG_OFF + 1;
-			CH->SLOT[s].evs = 0;
-		}
-	}
-}
-
-/* ----------  Create a virtual YM3812 ----------       */
-/* 'rate'  is sampling rate and 'bufsiz' is the size of the  */
-FM_OPL *OPLCreate(int type, int clock, int rate) {
-	char *ptr;
-	FM_OPL *OPL;
-	int state_size;
-	int max_ch = 9; /* normaly 9 channels */
-
-	if (OPL_LockTable() == -1)
-		return NULL;
-	/* allocate OPL state space */
-	state_size  = sizeof(FM_OPL);
-	state_size += sizeof(OPL_CH) * max_ch;
-
-	/* allocate memory block */
-	ptr = (char *)calloc(state_size, 1);
-	if (ptr == NULL)
-		return NULL;
-
-	/* clear */
-	memset(ptr, 0, state_size);
-	OPL       = (FM_OPL *)ptr; ptr += sizeof(FM_OPL);
-	OPL->P_CH = (OPL_CH *)ptr; ptr += sizeof(OPL_CH) * max_ch;
-
-	/* set channel state pointer */
-	OPL->type  = type;
-	OPL->clock = clock;
-	OPL->rate  = rate;
-	OPL->max_ch = max_ch;
-
-	/* init grobal tables */
-	OPL_initalize(OPL);
-
-	/* reset chip */
-	OPLResetChip(OPL);
-	return OPL;
-}
-
-/* ----------  Destroy one of vietual YM3812 ----------       */
-void OPLDestroy(FM_OPL *OPL) {
-	OPL_UnLockTable();
-	free(OPL);
-}
-
-/* ----------  Option handlers ----------       */
-void OPLSetTimerHandler(FM_OPL *OPL, OPL_TIMERHANDLER TimerHandler,int channelOffset) {
-	OPL->TimerHandler   = TimerHandler;
-	OPL->TimerParam = channelOffset;
-}
-
-void OPLSetIRQHandler(FM_OPL *OPL, OPL_IRQHANDLER IRQHandler, int param) {
-	OPL->IRQHandler     = IRQHandler;
-	OPL->IRQParam = param;
-}
-
-void OPLSetUpdateHandler(FM_OPL *OPL, OPL_UPDATEHANDLER UpdateHandler,int param) {
-	OPL->UpdateHandler = UpdateHandler;
-	OPL->UpdateParam = param;
-}
-
-/* ---------- YM3812 I/O interface ---------- */
-int OPLWrite(FM_OPL *OPL,int a,int v) {
-	if (!(a & 1)) {	/* address port */
-		OPL->address = v & 0xff;
-	} else {	/* data port */
-		if (OPL->UpdateHandler)
-			OPL->UpdateHandler(OPL->UpdateParam,0);
-		OPLWriteReg(OPL, OPL->address,v);
-	}
-	return OPL->status >> 7;
-}
-
-unsigned char OPLRead(FM_OPL *OPL,int a) {
-	if (!(a & 1)) {	/* status port */
-		return OPL->status & (OPL->statusmask | 0x80);
-	}
-	/* data port */
-	switch (OPL->address) {
-	case 0x05: /* KeyBoard IN */
-		warning("OPL:read unmapped KEYBOARD port");
-		return 0;
-	case 0x19: /* I/O DATA    */
-		warning("OPL:read unmapped I/O port");
-		return 0;
-	case 0x1a: /* PCM-DATA    */
-		return 0;
-	default:
-		break;
-	}
-	return 0;
-}
-
-int OPLTimerOver(FM_OPL *OPL, int c) {
-	if (c) {	/* Timer B */
-		OPL_STATUS_SET(OPL, 0x20);
-	} else {	/* Timer A */
-		OPL_STATUS_SET(OPL, 0x40);
-		/* CSM mode key,TL controll */
-		if (OPL->mode & 0x80) {	/* CSM mode total level latch and auto key on */
-			int ch;
-			if (OPL->UpdateHandler)
-				OPL->UpdateHandler(OPL->UpdateParam,0);
-			for (ch = 0; ch < 9; ch++)
-				CSMKeyControll(&OPL->P_CH[ch]);
-		}
-	}
-	/* reload timer */
-	if (OPL->TimerHandler)
-		(OPL->TimerHandler)(OPL->TimerParam + c, (double)OPL->T[c] * OPL->TimerBase);
-	return OPL->status >> 7;
-}
-
-FM_OPL *makeAdLibOPL(int rate) {
-	// We need to emulate one YM3812 chip
-	int env_bits = FMOPL_ENV_BITS_HQ;
-	int eg_ent = FMOPL_EG_ENT_HQ;
-#if defined (_WIN32_WCE) || defined(__SYMBIAN32__) || defined(__GP32__) || defined (GP2X) || defined(__MAEMO__) || defined(__DS__) || defined (__MINT__) || defined(__N64__)
-	if (ConfMan.hasKey("FM_high_quality") && ConfMan.getBool("FM_high_quality")) {
-		env_bits = FMOPL_ENV_BITS_HQ;
-		eg_ent = FMOPL_EG_ENT_HQ;
-	} else if (ConfMan.hasKey("FM_medium_quality") && ConfMan.getBool("FM_medium_quality")) {
-		env_bits = FMOPL_ENV_BITS_MQ;
-		eg_ent = FMOPL_EG_ENT_MQ;
-	} else {
-		env_bits = FMOPL_ENV_BITS_LQ;
-		eg_ent = FMOPL_EG_ENT_LQ;
-	}
-#endif
-
-	OPLBuildTables(env_bits, eg_ent);
-	return OPLCreate(OPL_TYPE_YM3812, 3579545, rate);
-}
-
-} // End of namespace MAME
-} // End of namespace OPL
-
diff --git a/sound/softsynth/opl/mame.h b/sound/softsynth/opl/mame.h
deleted file mode 100644
index ec812d0c4e..0000000000
--- a/sound/softsynth/opl/mame.h
+++ /dev/null
@@ -1,202 +0,0 @@
-/* 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.
- *
- * $URL$
- * $Id$
- *
- * LGPL licensed version of MAMEs fmopl (V0.37a modified) by
- * Tatsuyuki Satoh. Included from LGPL'ed AdPlug.
- */
-
-
-#ifndef SOUND_SOFTSYNTH_OPL_MAME_H
-#define SOUND_SOFTSYNTH_OPL_MAME_H
-
-#include "common/scummsys.h"
-#include "common/util.h"
-#include "common/random.h"
-
-#include "sound/fmopl.h"
-
-namespace OPL {
-namespace MAME {
-
-enum {
-	FMOPL_ENV_BITS_HQ = 16,
-	FMOPL_ENV_BITS_MQ = 8,
-	FMOPL_ENV_BITS_LQ = 8,
-	FMOPL_EG_ENT_HQ = 4096,
-	FMOPL_EG_ENT_MQ = 1024,
-	FMOPL_EG_ENT_LQ = 128
-};
-
-
-typedef void (*OPL_TIMERHANDLER)(int channel,double interval_Sec);
-typedef void (*OPL_IRQHANDLER)(int param,int irq);
-typedef void (*OPL_UPDATEHANDLER)(int param,int min_interval_us);
-
-#define OPL_TYPE_WAVESEL   0x01  /* waveform select    */
-
-/* Saving is necessary for member of the 'R' mark for suspend/resume */
-/* ---------- OPL one of slot  ---------- */
-typedef struct fm_opl_slot {
-	int TL;		/* total level     :TL << 8				*/
-	int TLL;	/* adjusted now TL						*/
-	uint8 KSR;	/* key scale rate  :(shift down bit)	*/
-	int *AR;	/* attack rate     :&AR_TABLE[AR<<2]	*/
-	int *DR;	/* decay rate      :&DR_TABLE[DR<<2]	*/
-	int SL;		/* sustain level   :SL_TABLE[SL]		*/
-	int *RR;	/* release rate    :&DR_TABLE[RR<<2]	*/
-	uint8 ksl;	/* keyscale level  :(shift down bits)	*/
-	uint8 ksr;	/* key scale rate  :kcode>>KSR			*/
-	uint mul;	/* multiple        :ML_TABLE[ML]		*/
-	uint Cnt;	/* frequency count						*/
-	uint Incr;	/* frequency step						*/
-
-	/* envelope generator state */
-	uint8 eg_typ;/* envelope type flag					*/
-	uint8 evm;	/* envelope phase						*/
-	int evc;	/* envelope counter						*/
-	int eve;	/* envelope counter end point			*/
-	int evs;	/* envelope counter step				*/
-	int evsa;	/* envelope step for AR :AR[ksr]		*/
-	int evsd;	/* envelope step for DR :DR[ksr]		*/
-	int evsr;	/* envelope step for RR :RR[ksr]		*/
-
-	/* LFO */
-	uint8 ams;		/* ams flag                            */
-	uint8 vib;		/* vibrate flag                        */
-	/* wave selector */
-	int **wavetable;
-} OPL_SLOT;
-
-/* ---------- OPL one of channel  ---------- */
-typedef struct fm_opl_channel {
-	OPL_SLOT SLOT[2];
-	uint8 CON;			/* connection type					*/
-	uint8 FB;			/* feed back       :(shift down bit)*/
-	int *connect1;		/* slot1 output pointer				*/
-	int *connect2;		/* slot2 output pointer				*/
-	int op1_out[2];		/* slot1 output for selfeedback		*/
-
-	/* phase generator state */
-	uint block_fnum;	/* block+fnum						*/
-	uint8 kcode;		/* key code        : KeyScaleCode	*/
-	uint fc;			/* Freq. Increment base				*/
-	uint ksl_base;		/* KeyScaleLevel Base step			*/
-	uint8 keyon;		/* key on/off flag					*/
-} OPL_CH;
-
-/* OPL state */
-typedef struct fm_opl_f {
-	uint8 type;			/* chip type                         */
-	int clock;			/* master clock  (Hz)                */
-	int rate;			/* sampling rate (Hz)                */
-	double freqbase;	/* frequency base                    */
-	double TimerBase;	/* Timer base time (==sampling time) */
-	uint8 address;		/* address register                  */
-	uint8 status;		/* status flag                       */
-	uint8 statusmask;	/* status mask                       */
-	uint mode;			/* Reg.08 : CSM , notesel,etc.       */
-
-	/* Timer */
-	int T[2];			/* timer counter                     */
-	uint8 st[2];		/* timer enable                      */
-
-	/* FM channel slots */
-	OPL_CH *P_CH;		/* pointer of CH                     */
-	int	max_ch;			/* maximum channel                   */
-
-	/* Rythm sention */
-	uint8 rythm;		/* Rythm mode , key flag */
-
-	/* time tables */
-	int AR_TABLE[76];	/* atttack rate tables				*/
-	int DR_TABLE[76];	/* decay rate tables				*/
-	uint FN_TABLE[1024];/* fnumber -> increment counter		*/
-
-	/* LFO */
-	int *ams_table;
-	int *vib_table;
-	int amsCnt;
-	int amsIncr;
-	int vibCnt;
-	int vibIncr;
-
-	/* wave selector enable flag */
-	uint8 wavesel;
-
-	/* external event callback handler */
-	OPL_TIMERHANDLER  TimerHandler;		/* TIMER handler   */
-	int TimerParam;						/* TIMER parameter */
-	OPL_IRQHANDLER    IRQHandler;		/* IRQ handler    */
-	int IRQParam;						/* IRQ parameter  */
-	OPL_UPDATEHANDLER UpdateHandler;	/* stream update handler   */
-	int UpdateParam;					/* stream update parameter */
-
-	Common::RandomSource rnd;
-} FM_OPL;
-
-/* ---------- Generic interface section ---------- */
-#define OPL_TYPE_YM3526 (0)
-#define OPL_TYPE_YM3812 (OPL_TYPE_WAVESEL)
-
-void OPLBuildTables(int ENV_BITS_PARAM, int EG_ENT_PARAM);
-
-FM_OPL *OPLCreate(int type, int clock, int rate);
-void OPLDestroy(FM_OPL *OPL);
-void OPLSetTimerHandler(FM_OPL *OPL, OPL_TIMERHANDLER TimerHandler, int channelOffset);
-void OPLSetIRQHandler(FM_OPL *OPL, OPL_IRQHANDLER IRQHandler, int param);
-void OPLSetUpdateHandler(FM_OPL *OPL, OPL_UPDATEHANDLER UpdateHandler, int param);
-
-void OPLResetChip(FM_OPL *OPL);
-int OPLWrite(FM_OPL *OPL, int a, int v);
-unsigned char OPLRead(FM_OPL *OPL, int a);
-int OPLTimerOver(FM_OPL *OPL, int c);
-void OPLWriteReg(FM_OPL *OPL, int r, int v);
-void YM3812UpdateOne(FM_OPL *OPL, int16 *buffer, int length);
-
-// Factory method
-FM_OPL *makeAdLibOPL(int rate);
-
-// OPL API implementation
-class OPL : public ::OPL::OPL {
-private:
-	FM_OPL *_opl;
-public:
-	OPL() : _opl(0) {}
-	~OPL();
-
-	bool init(int rate);
-	void reset();
-
-	void write(int a, int v);
-	byte read(int a);
-
-	void writeReg(int r, int v);
-
-	void readBuffer(int16 *buffer, int length);
-	bool isStereo() const { return false; }
-};
-
-} // End of namespace MAME
-} // End of namespace OPL
-
-#endif
diff --git a/sound/softsynth/pcspk.cpp b/sound/softsynth/pcspk.cpp
deleted file mode 100644
index bdf9f112e8..0000000000
--- a/sound/softsynth/pcspk.cpp
+++ /dev/null
@@ -1,187 +0,0 @@
-/* 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 "sound/softsynth/pcspk.h"
-#include "sound/null.h"
-
-namespace Audio {
-
-const PCSpeaker::generatorFunc PCSpeaker::generateWave[] =
-	{&PCSpeaker::generateSquare, &PCSpeaker::generateSine,
-	 &PCSpeaker::generateSaw,    &PCSpeaker::generateTriangle};
-
-PCSpeaker::PCSpeaker(int rate) {
-	_rate = rate;
-	_wave = kWaveFormSquare;
-	_playForever = false;
-	_oscLength = 0;
-	_oscSamples = 0;
-	_remainingSamples = 0;
-	_mixedSamples = 0;
-	_volume = 255;
-}
-
-PCSpeaker::~PCSpeaker() {
-}
-
-void PCSpeaker::play(WaveForm wave, int freq, int32 length) {
-	Common::StackLock lock(_mutex);
-
-	assert((wave >= kWaveFormSquare) && (wave <= kWaveFormTriangle));
-
-	_wave = wave;
-	_oscLength = _rate / freq;
-	_oscSamples = 0;
-	if (length == -1) {
-		_remainingSamples = 1;
-		_playForever = true;
-	} else {
-		_remainingSamples = (_rate * length) / 1000;
-		_playForever = false;
-	}
-	_mixedSamples = 0;
-}
-
-void PCSpeaker::stop(int32 delay) {
-	Common::StackLock lock(_mutex);
-
-	_remainingSamples = (_rate * delay) / 1000;
-	_playForever = false;
-}
-
-void PCSpeaker::setVolume(byte volume) {
-	_volume = volume;
-}
-
-bool PCSpeaker::isPlaying() const {
-	return _remainingSamples != 0;
-}
-
-int PCSpeaker::readBuffer(int16 *buffer, const int numSamples) {
-	Common::StackLock lock(_mutex);
-
-	int i;
-
-	for (i = 0; _remainingSamples && (i < numSamples); i++) {
-		buffer[i] = generateWave[_wave](_oscSamples, _oscLength) * _volume;
-		if (_oscSamples++ >= _oscLength)
-			_oscSamples = 0;
-		if (!_playForever)
-			_remainingSamples--;
-		_mixedSamples++;
-	}
-
-	// Clear the rest of the buffer
-	if (i < numSamples)
-		memset(buffer + i, 0, (numSamples - i) * sizeof(int16));
-
-	return numSamples;
-}
-
-int8 PCSpeaker::generateSquare(uint32 x, uint32 oscLength) {
-	return (x < (oscLength / 2)) ? 127 : -128;
-}
-
-int8 PCSpeaker::generateSine(uint32 x, uint32 oscLength) {
-	if (oscLength == 0)
-		return 0;
-
-	// TODO: Maybe using a look-up-table would be better?
-	return CLIP<int16>((int16) (128 * sin(2.0 * PI * x / oscLength)), -128, 127);
-}
-
-int8 PCSpeaker::generateSaw(uint32 x, uint32 oscLength) {
-	if (oscLength == 0)
-		return 0;
-
-	return ((x * (65536 / oscLength)) >> 8) - 128;
-}
-
-int8 PCSpeaker::generateTriangle(uint32 x, uint32 oscLength) {
-	if (oscLength == 0)
-		return 0;
-
-	int y = ((x * (65536 / (oscLength / 2))) >> 8) - 128;
-
-	return (x <= (oscLength / 2)) ? y : (256 - y);
-}
-
-} // End of namespace Audio
-
-
-//	Plugin interface
-//	(This can only create a null driver since pc speaker support 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 PCSpeakerMusicPlugin : public NullMusicPlugin {
-public:
-	const char *getName() const {
-		return _s("PC Speaker Emulator");
-	}
-
-	const char *getId() const {
-		return "pcspk";
-	}
-
-	MusicDevices getDevices() const;
-};
-
-MusicDevices PCSpeakerMusicPlugin::getDevices() const {
-	MusicDevices devices;
-	devices.push_back(MusicDevice(this, "", MT_PCSPK));
-	return devices;
-}
-
-class PCjrMusicPlugin : public NullMusicPlugin {
-public:
-	const char *getName() const {
-		return _s("IBM PCjr Emulator");
-	}
-
-	const char *getId() const {
-		return "pcjr";
-	}
-
-	MusicDevices getDevices() const;
-};
-
-MusicDevices PCjrMusicPlugin::getDevices() const {
-	MusicDevices devices;
-	devices.push_back(MusicDevice(this, "", MT_PCJR));
-	return devices;
-}
-
-//#if PLUGIN_ENABLED_DYNAMIC(PCSPK)
-	//REGISTER_PLUGIN_DYNAMIC(PCSPK, PLUGIN_TYPE_MUSIC, PCSpeakerMusicPlugin);
-//#else
-	REGISTER_PLUGIN_STATIC(PCSPK, PLUGIN_TYPE_MUSIC, PCSpeakerMusicPlugin);
-//#endif
-
-//#if PLUGIN_ENABLED_DYNAMIC(PCJR)
-	//REGISTER_PLUGIN_DYNAMIC(PCJR, PLUGIN_TYPE_MUSIC, PCjrMusicPlugin);
-//#else
-	REGISTER_PLUGIN_STATIC(PCJR, PLUGIN_TYPE_MUSIC, PCjrMusicPlugin);
-//#endif
diff --git a/sound/softsynth/pcspk.h b/sound/softsynth/pcspk.h
deleted file mode 100644
index 3cd8734a7e..0000000000
--- a/sound/softsynth/pcspk.h
+++ /dev/null
@@ -1,88 +0,0 @@
-/* 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_SOFTSYNTH_PCSPK_H
-#define SOUND_SOFTSYNTH_PCSPK_H
-
-#include "sound/audiostream.h"
-#include "sound/mixer.h"
-#include "common/mutex.h"
-
-namespace Audio {
-
-class PCSpeaker : public AudioStream {
-public:
-	enum WaveForm {
-		kWaveFormSquare = 0,
-		kWaveFormSine,
-		kWaveFormSaw,
-		kWaveFormTriangle
-	};
-
-	PCSpeaker(int rate = 44100);
-	~PCSpeaker();
-
-	/** Play a note for length ms.
-	 *
-	 *  If length is negative, play until told to stop.
-	 */
-	void play(WaveForm wave, int freq, int32 length);
-	/** Stop the currently playing note after delay ms. */
-	void stop(int32 delay = 0);
-	/** Adjust the volume. */
-	void setVolume(byte volume);
-
-	bool isPlaying() const;
-
-	int readBuffer(int16 *buffer, const int numSamples);
-
-	bool isStereo() const	{ return false; }
-	bool endOfData() const	{ return false; }
-	bool endOfStream() const { return false; }
-	int getRate() const	{ return _rate; }
-
-protected:
-	Common::Mutex _mutex;
-
-	int _rate;
-	WaveForm _wave;
-	bool _playForever;
-	uint32 _oscLength;
-	uint32 _oscSamples;
-	uint32 _remainingSamples;
-	uint32 _mixedSamples;
-	byte _volume;
-
-	typedef int8 (*generatorFunc)(uint32, uint32);
-	static const generatorFunc generateWave[];
-
-	static int8 generateSquare(uint32 x, uint32 oscLength);
-	static int8 generateSine(uint32 x, uint32 oscLength);
-	static int8 generateSaw(uint32 x, uint32 oscLength);
-	static int8 generateTriangle(uint32 x, uint32 oscLength);
-};
-
-} // End of namespace Audio
-
-#endif // SOUND_SOFTSYNTH_PCSPEAKER_H
diff --git a/sound/softsynth/sid.cpp b/sound/softsynth/sid.cpp
deleted file mode 100644
index c05ae2e8f5..0000000000
--- a/sound/softsynth/sid.cpp
+++ /dev/null
@@ -1,1456 +0,0 @@
-/* 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.
- *
- * $URL$
- * $Id$
- *
- */
-
-/*
- *  This file is based on reSID, a MOS6581 SID emulator engine.
- *  Copyright (C) 2004  Dag Lem <resid@nimrod.no>
- */
-
-#ifndef DISABLE_SID
-
-#include "sid.h"
-#include "sound/null.h"
-#include <math.h>
-
-namespace Resid {
-
-// Fixpoint constants (16.16 bits).
-const int SID::FIXP_SHIFT = 16;
-const int SID::FIXP_MASK = 0xffff;
-
-/*
- * WaveformGenerator
- */
-
-WaveformGenerator::WaveformGenerator() {
-	sync_source = this;
-
-	reset();
-}
-
-void WaveformGenerator::set_sync_source(WaveformGenerator* source) {
-	sync_source = source;
-	source->sync_dest = this;
-}
-
-void WaveformGenerator::writeFREQ_LO(reg8 freq_lo) {
-	freq = (freq & 0xff00) | (freq_lo & 0x00ff);
-}
-
-void WaveformGenerator::writeFREQ_HI(reg8 freq_hi) {
-	freq = ((freq_hi << 8) & 0xff00) | (freq & 0x00ff);
-}
-
-void WaveformGenerator::writePW_LO(reg8 pw_lo) {
-	pw = (pw & 0xf00) | (pw_lo & 0x0ff);
-}
-
-void WaveformGenerator::writePW_HI(reg8 pw_hi) {
-	pw = ((pw_hi << 8) & 0xf00) | (pw & 0x0ff);
-}
-
-void WaveformGenerator::writeCONTROL_REG(reg8 control) {
-	waveform = (control >> 4) & 0x0f;
-	ring_mod = control & 0x04;
-	sync = control & 0x02;
-
-	reg8 test_next = control & 0x08;
-
-	// Test bit set.
-	if (test_next) {
-		accumulator = 0;
-		shift_register = 0;
-	}
-	// Test bit cleared.
-	else if (test) {
-		shift_register = 0x7ffff8;
-	}
-
-	test = test_next;
-
-	// The gate bit is handled by the EnvelopeGenerator.
-}
-
-reg8 WaveformGenerator::readOSC() {
-	return output() >> 4;
-}
-
-void WaveformGenerator::reset() {
-	accumulator = 0;
-	shift_register = 0x7ffff8;
-	freq = 0;
-	pw = 0;
-
-	test = 0;
-	ring_mod = 0;
-	sync = 0;
-
-	msb_rising = false;
-}
-
-RESID_INLINE void WaveformGenerator::clock(cycle_count delta_t) {
-	// No operation if test bit is set.
-	if (test) {
-		return;
-	}
-
-	reg24 accumulator_prev = accumulator;
-
-	// Calculate new accumulator value;
-	reg24 delta_accumulator = delta_t*freq;
-	accumulator += delta_accumulator;
-	accumulator &= 0xffffff;
-
-	// Check whether the MSB is set high. This is used for synchronization.
-	msb_rising = !(accumulator_prev & 0x800000) && (accumulator & 0x800000);
-
-	// Shift noise register once for each time accumulator bit 19 is set high.
-	// Bit 19 is set high each time 2^20 (0x100000) is added to the accumulator.
-	reg24 shift_period = 0x100000;
-
-	while (delta_accumulator) {
-		if (delta_accumulator < shift_period) {
-			shift_period = delta_accumulator;
-			// Determine whether bit 19 is set on the last period.
-			// NB! Requires two's complement integer.
-			if (shift_period <= 0x080000) {
-				// Check for flip from 0 to 1.
-				if (((accumulator - shift_period) & 0x080000) || !(accumulator & 0x080000))
-				{
-					break;
-				}
-			}
-			else {
-				// Check for flip from 0 (to 1 or via 1 to 0) or from 1 via 0 to 1.
-				if (((accumulator - shift_period) & 0x080000) && !(accumulator & 0x080000))
-				{
-					break;
-				}
-			}
-		}
-
-		// Shift the noise/random register.
-		// NB! The shift is actually delayed 2 cycles, this is not modeled.
-		reg24 bit0 = ((shift_register >> 22) ^ (shift_register >> 17)) & 0x1;
-		shift_register <<= 1;
-		shift_register &= 0x7fffff;
-		shift_register |= bit0;
-
-		delta_accumulator -= shift_period;
-	}
-}
-
-
-/**
- * Synchronize oscillators.
- * This must be done after all the oscillators have been clock()'ed since the
- * oscillators operate in parallel.
- * Note that the oscillators must be clocked exactly on the cycle when the
- * MSB is set high for hard sync to operate correctly. See SID::clock().
- */
-RESID_INLINE void WaveformGenerator::synchronize() {
-	// A special case occurs when a sync source is synced itself on the same
-	// cycle as when its MSB is set high. In this case the destination will
-	// not be synced. This has been verified by sampling OSC3.
-	if (msb_rising && sync_dest->sync && !(sync && sync_source->msb_rising)) {
-		sync_dest->accumulator = 0;
-	}
-}
-
-
-/*
- * Output functions
- */
-
-// No waveform: Zero output.
-RESID_INLINE reg12 WaveformGenerator::output____() {
-	return 0x000;
-}
-
-// Triangle:
-RESID_INLINE reg12 WaveformGenerator::output___T() {
-	reg24 msb = (ring_mod ? accumulator ^ sync_source->accumulator : accumulator)
-		& 0x800000;
-	return ((msb ? ~accumulator : accumulator) >> 11) & 0xfff;
-}
-
-// Sawtooth:
-RESID_INLINE reg12 WaveformGenerator::output__S_() {
-	return accumulator >> 12;
-}
-
-// Pulse:
-RESID_INLINE reg12 WaveformGenerator::output_P__() {
-	return (test || (accumulator >> 12) >= pw) ? 0xfff : 0x000;
-}
-
-// Noise:
-RESID_INLINE reg12 WaveformGenerator::outputN___() {
-	return
-		((shift_register & 0x400000) >> 11) |
-		((shift_register & 0x100000) >> 10) |
-		((shift_register & 0x010000) >> 7) |
-		((shift_register & 0x002000) >> 5) |
-		((shift_register & 0x000800) >> 4) |
-		((shift_register & 0x000080) >> 1) |
-		((shift_register & 0x000010) << 1) |
-		((shift_register & 0x000004) << 2);
-}
-
-// Combined waveforms:
-
-RESID_INLINE reg12 WaveformGenerator::output__ST() {
-	return wave6581__ST[output__S_()] << 4;
-}
-
-RESID_INLINE reg12 WaveformGenerator::output_P_T() {
-	return (wave6581_P_T[output___T() >> 1] << 4) & output_P__();
-}
-
-RESID_INLINE reg12 WaveformGenerator::output_PS_() {
-	return (wave6581_PS_[output__S_()] << 4) & output_P__();
-}
-
-RESID_INLINE reg12 WaveformGenerator::output_PST() {
-	return (wave6581_PST[output__S_()] << 4) & output_P__();
-}
-
-// Combined waveforms including noise:
-
-RESID_INLINE reg12 WaveformGenerator::outputN__T() {
-	return 0;
-}
-
-RESID_INLINE reg12 WaveformGenerator::outputN_S_() {
-	return 0;
-}
-
-RESID_INLINE reg12 WaveformGenerator::outputN_ST() {
-	return 0;
-}
-
-RESID_INLINE reg12 WaveformGenerator::outputNP__() {
-	return 0;
-}
-
-RESID_INLINE reg12 WaveformGenerator::outputNP_T() {
-	return 0;
-}
-
-RESID_INLINE reg12 WaveformGenerator::outputNPS_() {
-	return 0;
-}
-
-RESID_INLINE reg12 WaveformGenerator::outputNPST() {
-	return 0;
-}
-
-/**
- * Select one of 16 possible combinations of waveforms.
- */
-RESID_INLINE reg12 WaveformGenerator::output() {
-	// It may seem cleaner to use an array of member functions to return
-	// waveform output; however a switch with inline functions is faster.
-
-	switch (waveform) {
-	default:
-	case 0x0:
-		return output____();
-	case 0x1:
-		return output___T();
-	case 0x2:
-		return output__S_();
-	case 0x3:
-		return output__ST();
-	case 0x4:
-		return output_P__();
-	case 0x5:
-		return output_P_T();
-	case 0x6:
-		return output_PS_();
-	case 0x7:
-		return output_PST();
-	case 0x8:
-		return outputN___();
-	case 0x9:
-		return outputN__T();
-	case 0xa:
-		return outputN_S_();
-	case 0xb:
-		return outputN_ST();
-	case 0xc:
-		return outputNP__();
-	case 0xd:
-		return outputNP_T();
-	case 0xe:
-		return outputNPS_();
-	case 0xf:
-		return outputNPST();
-	}
-}
-
-/*
- * Our objective is to construct a smooth interpolating single-valued function
- * y = f(x).
- * Our approach is to approximate the properties of Catmull-Rom splines for
- * piecewice cubic polynomials.
- */
-
-/**
- * Calculation of coefficients.
- */
-inline void cubic_coefficients(double x1, double y1, double x2, double y2,
-						double k1, double k2,
-						double& a, double& b, double& c, double& d)
-{
-	double dx = x2 - x1, dy = y2 - y1;
-
-	a = ((k1 + k2) - 2*dy/dx)/(dx*dx);
-	b = ((k2 - k1)/dx - 3*(x1 + x2)*a)/2;
-	c = k1 - (3*x1*a + 2*b)*x1;
-	d = y1 - ((x1*a + b)*x1 + c)*x1;
-}
-
-/**
- * Evaluation of cubic polynomial by forward differencing.
- */
-template<class PointPlotter>
-inline void interpolate_segment(double x1, double y1, double x2, double y2,
-								double k1, double k2,
-								PointPlotter plot, double res)
-{
-	double a, b, c, d;
-	cubic_coefficients(x1, y1, x2, y2, k1, k2, a, b, c, d);
-
-	double y = ((a*x1 + b)*x1 + c)*x1 + d;
-	double dy = (3*a*(x1 + res) + 2*b)*x1*res + ((a*res + b)*res + c)*res;
-	double d2y = (6*a*(x1 + res) + 2*b)*res*res;
-	double d3y = 6*a*res*res*res;
-
-	// Calculate each point.
-	for (double x = x1; x <= x2; x += res) {
-		plot(x, y);
-		y += dy; dy += d2y; d2y += d3y;
-	}
-}
-
-template<class PointIter>
-inline double x(PointIter p) {
-	return (*p)[0];
-}
-
-template<class PointIter>
-inline double y(PointIter p) {
-	return (*p)[1];
-}
-
-/**
- * Evaluation of complete interpolating function.
- * Note that since each curve segment is controlled by four points, the
- * end points will not be interpolated. If extra control points are not
- * desirable, the end points can simply be repeated to ensure interpolation.
- * Note also that points of non-differentiability and discontinuity can be
- * introduced by repeating points.
- */
-template<class PointIter, class PointPlotter>
-inline void interpolate(PointIter p0, PointIter pn, PointPlotter plot, double res) {
-	double k1, k2;
-
-	// Set up points for first curve segment.
-	PointIter p1 = p0; ++p1;
-	PointIter p2 = p1; ++p2;
-	PointIter p3 = p2; ++p3;
-
-	// Draw each curve segment.
-	for (; p2 != pn; ++p0, ++p1, ++p2, ++p3) {
-		// p1 and p2 equal; single point.
-		if (x(p1) == x(p2)) {
-			continue;
-		}
-		// Both end points repeated; straight line.
-		if (x(p0) == x(p1) && x(p2) == x(p3)) {
-			k1 = k2 = (y(p2) - y(p1))/(x(p2) - x(p1));
-		}
-		// p0 and p1 equal; use f''(x1) = 0.
-		else if (x(p0) == x(p1)) {
-			k2 = (y(p3) - y(p1))/(x(p3) - x(p1));
-			k1 = (3*(y(p2) - y(p1))/(x(p2) - x(p1)) - k2)/2;
-		}
-		// p2 and p3 equal; use f''(x2) = 0.
-		else if (x(p2) == x(p3)) {
-			k1 = (y(p2) - y(p0))/(x(p2) - x(p0));
-			k2 = (3*(y(p2) - y(p1))/(x(p2) - x(p1)) - k1)/2;
-		}
-		// Normal curve.
-		else {
-			k1 = (y(p2) - y(p0))/(x(p2) - x(p0));
-			k2 = (y(p3) - y(p1))/(x(p3) - x(p1));
-		}
-
-		interpolate_segment(x(p1), y(p1), x(p2), y(p2), k1, k2, plot, res);
-	}
-}
-
-/**
- * Class for plotting integers into an array.
- */
-template<class F>
-class PointPlotter {
-protected:
-	F* f;
-
-public:
-	PointPlotter(F* arr) : f(arr) {
-	}
-
-	void operator ()(double x, double y) {
-		// Clamp negative values to zero.
-		if (y < 0) {
-			y = 0;
-		}
-
-		f[F(x)] = F(y);
-	}
-};
-
-fc_point Filter::f0_points_6581[] = {
-	//  FC      f         FCHI FCLO
-	// ----------------------------
-	{    0,   220 },   // 0x00      - repeated end point
-	{    0,   220 },   // 0x00
-	{  128,   230 },   // 0x10
-	{  256,   250 },   // 0x20
-	{  384,   300 },   // 0x30
-	{  512,   420 },   // 0x40
-	{  640,   780 },   // 0x50
-	{  768,  1600 },   // 0x60
-	{  832,  2300 },   // 0x68
-	{  896,  3200 },   // 0x70
-	{  960,  4300 },   // 0x78
-	{  992,  5000 },   // 0x7c
-	{ 1008,  5400 },   // 0x7e
-	{ 1016,  5700 },   // 0x7f
-	{ 1023,  6000 },   // 0x7f 0x07
-	{ 1023,  6000 },   // 0x7f 0x07 - discontinuity
-	{ 1024,  4600 },   // 0x80      -
-	{ 1024,  4600 },   // 0x80
-	{ 1032,  4800 },   // 0x81
-	{ 1056,  5300 },   // 0x84
-	{ 1088,  6000 },   // 0x88
-	{ 1120,  6600 },   // 0x8c
-	{ 1152,  7200 },   // 0x90
-	{ 1280,  9500 },   // 0xa0
-	{ 1408, 12000 },   // 0xb0
-	{ 1536, 14500 },   // 0xc0
-	{ 1664, 16000 },   // 0xd0
-	{ 1792, 17100 },   // 0xe0
-	{ 1920, 17700 },   // 0xf0
-	{ 2047, 18000 },   // 0xff 0x07
-	{ 2047, 18000 }    // 0xff 0x07 - repeated end point
-};
-
-
-/*
- * Filter
- */
-
-Filter::Filter() {
-	fc = 0;
-
-	res = 0;
-
-	filt = 0;
-
-	voice3off = 0;
-
-	hp_bp_lp = 0;
-
-	vol = 0;
-
-	// State of filter.
-	Vhp = 0;
-	Vbp = 0;
-	Vlp = 0;
-	Vnf = 0;
-
-	enable_filter(true);
-
-	// Create mappings from FC to cutoff frequency.
-	interpolate(f0_points_6581, f0_points_6581
-		+ sizeof(f0_points_6581)/sizeof(*f0_points_6581) - 1,
-		PointPlotter<sound_sample>(f0_6581), 1.0);
-
-	mixer_DC = (-0xfff*0xff/18) >> 7;
-
-	f0 = f0_6581;
-	f0_points = f0_points_6581;
-	f0_count = sizeof(f0_points_6581)/sizeof(*f0_points_6581);
-
-	set_w0();
-	set_Q();
-}
-
-void Filter::enable_filter(bool enable) {
-	enabled = enable;
-}
-
-void Filter::reset(){
-	fc = 0;
-
-	res = 0;
-
-	filt = 0;
-
-	voice3off = 0;
-
-	hp_bp_lp = 0;
-
-	vol = 0;
-
-	// State of filter.
-	Vhp = 0;
-	Vbp = 0;
-	Vlp = 0;
-	Vnf = 0;
-
-	set_w0();
-	set_Q();
-}
-
-void Filter::writeFC_LO(reg8 fc_lo) {
-	fc = (fc & 0x7f8) | (fc_lo & 0x007);
-	set_w0();
-}
-
-void Filter::writeFC_HI(reg8 fc_hi) {
-	fc = ((fc_hi << 3) & 0x7f8) | (fc & 0x007);
-	set_w0();
-}
-
-void Filter::writeRES_FILT(reg8 res_filt) {
-	res = (res_filt >> 4) & 0x0f;
-	set_Q();
-
-	filt = res_filt & 0x0f;
-}
-
-void Filter::writeMODE_VOL(reg8 mode_vol) {
-	voice3off = mode_vol & 0x80;
-
-	hp_bp_lp = (mode_vol >> 4) & 0x07;
-
-	vol = mode_vol & 0x0f;
-}
-
-// Set filter cutoff frequency.
-void Filter::set_w0() {
-	const double pi = 3.1415926535897932385;
-
-	// Multiply with 1.048576 to facilitate division by 1 000 000 by right-
-	// shifting 20 times (2 ^ 20 = 1048576).
-	w0 = static_cast<sound_sample>(2*pi*f0[fc]*1.048576);
-
-	// Limit f0 to 16kHz to keep 1 cycle filter stable.
-	const sound_sample w0_max_1 = static_cast<sound_sample>(2*pi*16000*1.048576);
-	w0_ceil_1 = w0 <= w0_max_1 ? w0 : w0_max_1;
-
-	// Limit f0 to 4kHz to keep delta_t cycle filter stable.
-	const sound_sample w0_max_dt = static_cast<sound_sample>(2*pi*4000*1.048576);
-	w0_ceil_dt = w0 <= w0_max_dt ? w0 : w0_max_dt;
-}
-
-// Set filter resonance.
-void Filter::set_Q() {
-	// Q is controlled linearly by res. Q has approximate range [0.707, 1.7].
-	// As resonance is increased, the filter must be clocked more often to keep
-	// stable.
-
-	// The coefficient 1024 is dispensed of later by right-shifting 10 times
-	// (2 ^ 10 = 1024).
-	_1024_div_Q = static_cast<sound_sample>(1024.0/(0.707 + 1.0*res/0x0f));
-}
-
-RESID_INLINE void Filter::clock(cycle_count delta_t,
-				   sound_sample voice1,
-				   sound_sample voice2,
-				   sound_sample voice3)
-{
-	// Scale each voice down from 20 to 13 bits.
-	voice1 >>= 7;
-	voice2 >>= 7;
-
-	// NB! Voice 3 is not silenced by voice3off if it is routed through
-	// the filter.
-	if (voice3off && !(filt & 0x04)) {
-		voice3 = 0;
-	}
-	else {
-		voice3 >>= 7;
-	}
-
-	// Enable filter on/off.
-	// This is not really part of SID, but is useful for testing.
-	// On slow CPUs it may be necessary to bypass the filter to lower the CPU
-	// load.
-	if (!enabled) {
-		Vnf = voice1 + voice2 + voice3;
-		Vhp = Vbp = Vlp = 0;
-		return;
-	}
-
-	// Route voices into or around filter.
-	// The code below is expanded to a switch for faster execution.
-	// (filt1 ? Vi : Vnf) += voice1;
-	// (filt2 ? Vi : Vnf) += voice2;
-	// (filt3 ? Vi : Vnf) += voice3;
-
-	sound_sample Vi;
-
-	switch (filt) {
-	default:
-	case 0x0:
-		Vi = 0;
-		Vnf = voice1 + voice2 + voice3;
-		break;
-	case 0x1:
-		Vi = voice1;
-		Vnf = voice2 + voice3;
-		break;
-	case 0x2:
-		Vi = voice2;
-		Vnf = voice1 + voice3;
-		break;
-	case 0x3:
-		Vi = voice1 + voice2;
-		Vnf = voice3;
-		break;
-	case 0x4:
-		Vi = voice3;
-		Vnf = voice1 + voice2;
-		break;
-	case 0x5:
-		Vi = voice1 + voice3;
-		Vnf = voice2;
-		break;
-	case 0x6:
-		Vi = voice2 + voice3;
-		Vnf = voice1;
-		break;
-	case 0x7:
-		Vi = voice1 + voice2 + voice3;
-		Vnf = 0;
-		break;
-	case 0x8:
-		Vi = 0;
-		Vnf = voice1 + voice2 + voice3;
-		break;
-	case 0x9:
-		Vi = voice1;
-		Vnf = voice2 + voice3;
-		break;
-	case 0xa:
-		Vi = voice2;
-		Vnf = voice1 + voice3;
-		break;
-	case 0xb:
-		Vi = voice1 + voice2;
-		Vnf = voice3;
-		break;
-	case 0xc:
-		Vi = voice3;
-		Vnf = voice1 + voice2;
-		break;
-	case 0xd:
-		Vi = voice1 + voice3;
-		Vnf = voice2;
-		break;
-	case 0xe:
-		Vi = voice2 + voice3;
-		Vnf = voice1;
-		break;
-	case 0xf:
-		Vi = voice1 + voice2 + voice3;
-		Vnf = 0;
-		break;
-	}
-
-	// Maximum delta cycles for the filter to work satisfactorily under current
-	// cutoff frequency and resonance constraints is approximately 8.
-	cycle_count delta_t_flt = 8;
-
-	while (delta_t) {
-		if (delta_t < delta_t_flt) {
-			delta_t_flt = delta_t;
-		}
-
-		// delta_t is converted to seconds given a 1MHz clock by dividing
-		// with 1 000 000. This is done in two operations to avoid integer
-		// multiplication overflow.
-
-		// Calculate filter outputs.
-		// Vhp = Vbp/Q - Vlp - Vi;
-		// dVbp = -w0*Vhp*dt;
-		// dVlp = -w0*Vbp*dt;
-		sound_sample w0_delta_t = w0_ceil_dt*delta_t_flt >> 6;
-
-		sound_sample dVbp = (w0_delta_t*Vhp >> 14);
-		sound_sample dVlp = (w0_delta_t*Vbp >> 14);
-		Vbp -= dVbp;
-		Vlp -= dVlp;
-		Vhp = (Vbp*_1024_div_Q >> 10) - Vlp - Vi;
-
-		delta_t -= delta_t_flt;
-	}
-}
-
-RESID_INLINE sound_sample Filter::output() {
-	// This is handy for testing.
-	if (!enabled) {
-		return (Vnf + mixer_DC)*static_cast<sound_sample>(vol);
-	}
-
-	// Mix highpass, bandpass, and lowpass outputs. The sum is not
-	// weighted, this can be confirmed by sampling sound output for
-	// e.g. bandpass, lowpass, and bandpass+lowpass from a SID chip.
-
-	// The code below is expanded to a switch for faster execution.
-	// if (hp) Vf += Vhp;
-	// if (bp) Vf += Vbp;
-	// if (lp) Vf += Vlp;
-
-	sound_sample Vf;
-
-	switch (hp_bp_lp) {
-	default:
-	case 0x0:
-		Vf = 0;
-		break;
-	case 0x1:
-		Vf = Vlp;
-		break;
-	case 0x2:
-		Vf = Vbp;
-		break;
-	case 0x3:
-		Vf = Vlp + Vbp;
-		break;
-	case 0x4:
-		Vf = Vhp;
-		break;
-	case 0x5:
-		Vf = Vlp + Vhp;
-		break;
-	case 0x6:
-		Vf = Vbp + Vhp;
-		break;
-	case 0x7:
-		Vf = Vlp + Vbp + Vhp;
-		break;
-	}
-
-	// Sum non-filtered and filtered output.
-	// Multiply the sum with volume.
-	return (Vnf + Vf + mixer_DC)*static_cast<sound_sample>(vol);
-}
-
-
-/*
- * EnvelopeGenerator
- */
-
-EnvelopeGenerator::EnvelopeGenerator() {
-	reset();
-}
-
-void EnvelopeGenerator::reset() {
-	envelope_counter = 0;
-
-	attack = 0;
-	decay = 0;
-	sustain = 0;
-	release = 0;
-
-	gate = 0;
-
-	rate_counter = 0;
-	exponential_counter = 0;
-	exponential_counter_period = 1;
-
-	state = RELEASE;
-	rate_period = rate_counter_period[release];
-	hold_zero = true;
-}
-
-reg16 EnvelopeGenerator::rate_counter_period[] = {
-	9,  //   2ms*1.0MHz/256 =     7.81
-	32,  //   8ms*1.0MHz/256 =    31.25
-	63,  //  16ms*1.0MHz/256 =    62.50
-	95,  //  24ms*1.0MHz/256 =    93.75
-	149,  //  38ms*1.0MHz/256 =   148.44
-	220,  //  56ms*1.0MHz/256 =   218.75
-	267,  //  68ms*1.0MHz/256 =   265.63
-	313,  //  80ms*1.0MHz/256 =   312.50
-	392,  // 100ms*1.0MHz/256 =   390.63
-	977,  // 250ms*1.0MHz/256 =   976.56
-	1954,  // 500ms*1.0MHz/256 =  1953.13
-	3126,  // 800ms*1.0MHz/256 =  3125.00
-	3907,  //   1 s*1.0MHz/256 =  3906.25
-	11720,  //   3 s*1.0MHz/256 = 11718.75
-	19532,  //   5 s*1.0MHz/256 = 19531.25
-	31251   //   8 s*1.0MHz/256 = 31250.00
-};
-
-
-reg8 EnvelopeGenerator::sustain_level[] = {
-	0x00,
-	0x11,
-	0x22,
-	0x33,
-	0x44,
-	0x55,
-	0x66,
-	0x77,
-	0x88,
-	0x99,
-	0xaa,
-	0xbb,
-	0xcc,
-	0xdd,
-	0xee,
-	0xff,
-};
-
-void EnvelopeGenerator::writeCONTROL_REG(reg8 control) {
-	reg8 gate_next = control & 0x01;
-
-	// The rate counter is never reset, thus there will be a delay before the
-	// envelope counter starts counting up (attack) or down (release).
-
-	// Gate bit on: Start attack, decay, sustain.
-	if (!gate && gate_next) {
-		state = ATTACK;
-		rate_period = rate_counter_period[attack];
-
-		// Switching to attack state unlocks the zero freeze.
-		hold_zero = false;
-	}
-	// Gate bit off: Start release.
-	else if (gate && !gate_next) {
-		state = RELEASE;
-		rate_period = rate_counter_period[release];
-	}
-
-	gate = gate_next;
-}
-
-void EnvelopeGenerator::writeATTACK_DECAY(reg8 attack_decay) {
-	attack = (attack_decay >> 4) & 0x0f;
-	decay = attack_decay & 0x0f;
-	if (state == ATTACK) {
-		rate_period = rate_counter_period[attack];
-	}
-	else if (state == DECAY_SUSTAIN) {
-		rate_period = rate_counter_period[decay];
-	}
-}
-
-void EnvelopeGenerator::writeSUSTAIN_RELEASE(reg8 sustain_release) {
-	sustain = (sustain_release >> 4) & 0x0f;
-	release = sustain_release & 0x0f;
-	if (state == RELEASE) {
-		rate_period = rate_counter_period[release];
-	}
-}
-
-reg8 EnvelopeGenerator::readENV() {
-	return output();
-}
-
-RESID_INLINE void EnvelopeGenerator::clock(cycle_count delta_t) {
-	// Check for ADSR delay bug.
-	// If the rate counter comparison value is set below the current value of the
-	// rate counter, the counter will continue counting up until it wraps around
-	// to zero at 2^15 = 0x8000, and then count rate_period - 1 before the
-	// envelope can finally be stepped.
-	// This has been verified by sampling ENV3.
-	//
-
-	// NB! This requires two's complement integer.
-	int rate_step = rate_period - rate_counter;
-	if (rate_step <= 0) {
-		rate_step += 0x7fff;
-	}
-
-	while (delta_t) {
-		if (delta_t < rate_step) {
-			rate_counter += delta_t;
-			if (rate_counter & 0x8000) {
-				++rate_counter &= 0x7fff;
-			}
-			return;
-		}
-
-		rate_counter = 0;
-		delta_t -= rate_step;
-
-		// The first envelope step in the attack state also resets the exponential
-		// counter. This has been verified by sampling ENV3.
-		//
-		if (state == ATTACK	|| ++exponential_counter == exponential_counter_period)
-		{
-			exponential_counter = 0;
-
-			// Check whether the envelope counter is frozen at zero.
-			if (hold_zero) {
-				rate_step = rate_period;
-				continue;
-			}
-
-			switch (state) {
-			case ATTACK:
-				// The envelope counter can flip from 0xff to 0x00 by changing state to
-				// release, then to attack. The envelope counter is then frozen at
-				// zero; to unlock this situation the state must be changed to release,
-				// then to attack. This has been verified by sampling ENV3.
-				//
-				++envelope_counter &= 0xff;
-				if (envelope_counter == 0xff) {
-					state = DECAY_SUSTAIN;
-					rate_period = rate_counter_period[decay];
-				}
-				break;
-			case DECAY_SUSTAIN:
-				if (envelope_counter != sustain_level[sustain]) {
-					--envelope_counter;
-				}
-				break;
-			case RELEASE:
-				// The envelope counter can flip from 0x00 to 0xff by changing state to
-				// attack, then to release. The envelope counter will then continue
-				// counting down in the release state.
-				// This has been verified by sampling ENV3.
-				// NB! The operation below requires two's complement integer.
-				//
-				--envelope_counter &= 0xff;
-				break;
-			}
-
-			// Check for change of exponential counter period.
-			switch (envelope_counter) {
-			case 0xff:
-				exponential_counter_period = 1;
-				break;
-			case 0x5d:
-				exponential_counter_period = 2;
-				break;
-			case 0x36:
-				exponential_counter_period = 4;
-				break;
-			case 0x1a:
-				exponential_counter_period = 8;
-				break;
-			case 0x0e:
-				exponential_counter_period = 16;
-				break;
-			case 0x06:
-				exponential_counter_period = 30;
-				break;
-			case 0x00:
-				exponential_counter_period = 1;
-
-				// When the envelope counter is changed to zero, it is frozen at zero.
-				// This has been verified by sampling ENV3.
-				hold_zero = true;
-				break;
-			}
-		}
-
-		rate_step = rate_period;
-	}
-}
-
-RESID_INLINE reg8 EnvelopeGenerator::output() {
-	return envelope_counter;
-}
-
-
-/*
- * ExternalFilter
- */
-
-ExternalFilter::ExternalFilter() {
-	reset();
-	enable_filter(true);
-	set_sampling_parameter(15915.6);
-	mixer_DC = ((((0x800 - 0x380) + 0x800)*0xff*3 - 0xfff*0xff/18) >> 7)*0x0f;
-}
-
-void ExternalFilter::enable_filter(bool enable) {
-	enabled = enable;
-}
-
-void ExternalFilter::set_sampling_parameter(double pass_freq) {
-	static const double pi = 3.1415926535897932385;
-
-	w0hp = 105;
-	w0lp = (sound_sample) (pass_freq * (2.0 * pi * 1.048576));
-	if (w0lp > 104858)
-		w0lp = 104858;
-}
-
-void ExternalFilter::reset() {
-	// State of filter.
-	Vlp = 0;
-	Vhp = 0;
-	Vo = 0;
-}
-
-RESID_INLINE void ExternalFilter::clock(cycle_count delta_t, sound_sample Vi) {
-	// This is handy for testing.
-	if (!enabled) {
-		// Remove maximum DC level since there is no filter to do it.
-		Vlp = Vhp = 0;
-		Vo = Vi - mixer_DC;
-		return;
-	}
-
-	// Maximum delta cycles for the external filter to work satisfactorily
-	// is approximately 8.
-	cycle_count delta_t_flt = 8;
-
-	while (delta_t) {
-		if (delta_t < delta_t_flt) {
-			delta_t_flt = delta_t;
-		}
-
-		// delta_t is converted to seconds given a 1MHz clock by dividing
-		// with 1 000 000.
-
-		// Calculate filter outputs.
-		// Vo  = Vlp - Vhp;
-		// Vlp = Vlp + w0lp*(Vi - Vlp)*delta_t;
-		// Vhp = Vhp + w0hp*(Vlp - Vhp)*delta_t;
-
-		sound_sample dVlp = (w0lp*delta_t_flt >> 8)*(Vi - Vlp) >> 12;
-		sound_sample dVhp = w0hp*delta_t_flt*(Vlp - Vhp) >> 20;
-		Vo = Vlp - Vhp;
-		Vlp += dVlp;
-		Vhp += dVhp;
-
-		delta_t -= delta_t_flt;
-	}
-}
-
-RESID_INLINE sound_sample ExternalFilter::output() {
-	return Vo;
-}
-
-
-/*
- * Voice
- */
-
-Voice::Voice() {
-	wave_zero = 0x380;
-	voice_DC = 0x800*0xff;
-}
-
-void Voice::set_sync_source(Voice* source) {
-	wave.set_sync_source(&source->wave);
-}
-
-void Voice::writeCONTROL_REG(reg8 control) {
-	wave.writeCONTROL_REG(control);
-	envelope.writeCONTROL_REG(control);
-}
-
-void Voice::reset() {
-	wave.reset();
-	envelope.reset();
-}
-
-
-/*
- * SID
- */
-
-SID::SID() {
-	voice[0].set_sync_source(&voice[2]);
-	voice[1].set_sync_source(&voice[0]);
-	voice[2].set_sync_source(&voice[1]);
-
-	set_sampling_parameters(985248, 44100);
-
-	bus_value = 0;
-	bus_value_ttl = 0;
-}
-
-SID::~SID() {}
-
-void SID::reset() {
-	for (int i = 0; i < 3; i++) {
-		voice[i].reset();
-	}
-	filter.reset();
-	extfilt.reset();
-
-	bus_value = 0;
-	bus_value_ttl = 0;
-}
-
-int SID::output() {
-	const int range = 1 << 16;
-	const int half = range >> 1;
-	int sample = extfilt.output()/((4095*255 >> 7)*3*15*2/range);
-	if (sample >= half) {
-		return half - 1;
-	}
-	if (sample < -half) {
-		return -half;
-	}
-	return sample;
-}
-
-
-/**
- * Read registers.
- *
- * Reading a write only register returns the last byte written to any SID
- * register. The individual bits in this value start to fade down towards
- * zero after a few cycles. All bits reach zero within approximately
- * $2000 - $4000 cycles.
- * It has been claimed that this fading happens in an orderly fashion, however
- * sampling of write only registers reveals that this is not the case.
- * NB! This is not correctly modeled.
- * The actual use of write only registers has largely been made in the belief
- * that all SID registers are readable. To support this belief the read
- * would have to be done immediately after a write to the same register
- * (remember that an intermediate write to another register would yield that
- * value instead). With this in mind we return the last value written to
- * any SID register for $2000 cycles without modeling the bit fading.
- */
-reg8 SID::read(reg8 offset) {
-	switch (offset) {
-		case 0x19:
-		case 0x1a:
-			return 0; //readPOT();
-		case 0x1b:
-			return voice[2].wave.readOSC();
-		case 0x1c:
-			return voice[2].envelope.readENV();
-		default:
-			return bus_value;
-	}
-}
-
-void SID::write(reg8 offset, reg8 value) {
-	bus_value = value;
-	bus_value_ttl = 0x2000;
-
-	switch (offset) {
-	  case 0x00:
-		  voice[0].wave.writeFREQ_LO(value);
-		  break;
-	  case 0x01:
-		  voice[0].wave.writeFREQ_HI(value);
-		  break;
-	  case 0x02:
-		  voice[0].wave.writePW_LO(value);
-		  break;
-	  case 0x03:
-		  voice[0].wave.writePW_HI(value);
-		  break;
-	  case 0x04:
-		  voice[0].writeCONTROL_REG(value);
-		  break;
-	  case 0x05:
-		  voice[0].envelope.writeATTACK_DECAY(value);
-		  break;
-	  case 0x06:
-		  voice[0].envelope.writeSUSTAIN_RELEASE(value);
-		  break;
-	  case 0x07:
-		  voice[1].wave.writeFREQ_LO(value);
-		  break;
-	  case 0x08:
-		  voice[1].wave.writeFREQ_HI(value);
-		  break;
-	  case 0x09:
-		  voice[1].wave.writePW_LO(value);
-		  break;
-	  case 0x0a:
-		  voice[1].wave.writePW_HI(value);
-		  break;
-	  case 0x0b:
-		  voice[1].writeCONTROL_REG(value);
-		  break;
-	  case 0x0c:
-		  voice[1].envelope.writeATTACK_DECAY(value);
-		  break;
-	  case 0x0d:
-		  voice[1].envelope.writeSUSTAIN_RELEASE(value);
-		  break;
-	  case 0x0e:
-		  voice[2].wave.writeFREQ_LO(value);
-		  break;
-	  case 0x0f:
-		  voice[2].wave.writeFREQ_HI(value);
-		  break;
-	  case 0x10:
-		  voice[2].wave.writePW_LO(value);
-		  break;
-	  case 0x11:
-		  voice[2].wave.writePW_HI(value);
-		  break;
-	  case 0x12:
-		  voice[2].writeCONTROL_REG(value);
-		  break;
-	  case 0x13:
-		  voice[2].envelope.writeATTACK_DECAY(value);
-		  break;
-	  case 0x14:
-		  voice[2].envelope.writeSUSTAIN_RELEASE(value);
-		  break;
-	  case 0x15:
-		  filter.writeFC_LO(value);
-		  break;
-	  case 0x16:
-		  filter.writeFC_HI(value);
-		  break;
-	  case 0x17:
-		  filter.writeRES_FILT(value);
-		  break;
-	  case 0x18:
-		  filter.writeMODE_VOL(value);
-		  break;
-	  default:
-		  break;
-	}
-}
-
-void SID::enable_filter(bool enable) {
-	filter.enable_filter(enable);
-}
-
-void SID::enable_external_filter(bool enable) {
-	extfilt.enable_filter(enable);
-}
-
-
-/**
- * Setting of SID sampling parameters.
- *
- * Use a clock freqency of 985248Hz for PAL C64, 1022730Hz for NTSC C64.
- * The default end of passband frequency is pass_freq = 0.9*sample_freq/2
- * for sample frequencies up to ~ 44.1kHz, and 20kHz for higher sample
- * frequencies.
- *
- * For resampling, the ratio between the clock frequency and the sample
- * frequency is limited as follows:
- *   125*clock_freq/sample_freq < 16384
- * E.g. provided a clock frequency of ~ 1MHz, the sample frequency can not
- * be set lower than ~ 8kHz. A lower sample frequency would make the
- * resampling code overfill its 16k sample ring buffer.
- *
- * The end of passband frequency is also limited:
- *   pass_freq <= 0.9*sample_freq/2
- *
- * E.g. for a 44.1kHz sampling rate the end of passband frequency is limited
- * to slightly below 20kHz. This constraint ensures that the FIR table is
- * not overfilled.
- */
-bool SID::set_sampling_parameters(double clock_freq,
-								  double sample_freq, double pass_freq,
-								  double filter_scale)
-{
-	// The default passband limit is 0.9*sample_freq/2 for sample
-	// frequencies below ~ 44.1kHz, and 20kHz for higher sample frequencies.
-	if (pass_freq < 0) {
-		pass_freq = 20000;
-		if (2*pass_freq/sample_freq >= 0.9) {
-			pass_freq = 0.9*sample_freq/2;
-		}
-	}
-	// Check whether the FIR table would overfill.
-	else if (pass_freq > 0.9*sample_freq/2) {
-		return false;
-	}
-
-	// The filter scaling is only included to avoid clipping, so keep
-	// it sane.
-	if (filter_scale < 0.9 || filter_scale > 1.0) {
-		return false;
-	}
-
-	// Set the external filter to the pass freq
-	extfilt.set_sampling_parameter (pass_freq);
-	clock_frequency = clock_freq;
-
-	cycles_per_sample =
-		cycle_count(clock_freq/sample_freq*(1 << FIXP_SHIFT) + 0.5);
-
-	sample_offset = 0;
-	sample_prev = 0;
-
-	return true;
-}
-
-void SID::clock(cycle_count delta_t) {
-	int i;
-
-	if (delta_t <= 0) {
-		return;
-	}
-
-	// Age bus value.
-	bus_value_ttl -= delta_t;
-	if (bus_value_ttl <= 0) {
-		bus_value = 0;
-		bus_value_ttl = 0;
-	}
-
-	// Clock amplitude modulators.
-	for (i = 0; i < 3; i++) {
-		voice[i].envelope.clock(delta_t);
-	}
-
-	// Clock and synchronize oscillators.
-	// Loop until we reach the current cycle.
-	cycle_count delta_t_osc = delta_t;
-	while (delta_t_osc) {
-		cycle_count delta_t_min = delta_t_osc;
-
-		// Find minimum number of cycles to an oscillator accumulator MSB toggle.
-		// We have to clock on each MSB on / MSB off for hard sync to operate
-		// correctly.
-		for (i = 0; i < 3; i++) {
-			WaveformGenerator& wave = voice[i].wave;
-
-			// It is only necessary to clock on the MSB of an oscillator that is
-			// a sync source and has freq != 0.
-			if (!(wave.sync_dest->sync && wave.freq)) {
-				continue;
-			}
-
-			reg16 freq = wave.freq;
-			reg24 accumulator = wave.accumulator;
-
-			// Clock on MSB off if MSB is on, clock on MSB on if MSB is off.
-			reg24 delta_accumulator =
-				(accumulator & 0x800000 ? 0x1000000 : 0x800000) - accumulator;
-
-			cycle_count delta_t_next = delta_accumulator/freq;
-			if (delta_accumulator%freq) {
-				++delta_t_next;
-			}
-
-			if (delta_t_next < delta_t_min) {
-				delta_t_min = delta_t_next;
-			}
-		}
-
-		// Clock oscillators.
-		for (i = 0; i < 3; i++) {
-			voice[i].wave.clock(delta_t_min);
-		}
-
-		// Synchronize oscillators.
-		for (i = 0; i < 3; i++) {
-			voice[i].wave.synchronize();
-		}
-
-		delta_t_osc -= delta_t_min;
-	}
-
-	// Clock filter.
-	filter.clock(delta_t,
-		voice[0].output(), voice[1].output(), voice[2].output());
-
-	// Clock external filter.
-	extfilt.clock(delta_t, filter.output());
-}
-
-
-/**
- * SID clocking with audio sampling.
- * Fixpoint arithmetics is used.
- */
-int SID::clock(cycle_count& delta_t, short* buf, int n, int interleave) {
-	int s = 0;
-
-	for (;;) {
-		cycle_count next_sample_offset = sample_offset + cycles_per_sample + (1 << (FIXP_SHIFT - 1));
-		cycle_count delta_t_sample = next_sample_offset >> FIXP_SHIFT;
-		if (delta_t_sample > delta_t) {
-			break;
-		}
-		if (s >= n) {
-			return s;
-		}
-		clock(delta_t_sample);
-		delta_t -= delta_t_sample;
-		sample_offset = (next_sample_offset & FIXP_MASK) - (1 << (FIXP_SHIFT - 1));
-		buf[s++*interleave] = output();
-	}
-
-	clock(delta_t);
-	sample_offset -= delta_t << FIXP_SHIFT;
-	delta_t = 0;
-	return s;
-}
-
-}
-
-//	Plugin interface
-//	(This can only create a null driver since C64 audio support 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 C64MusicPlugin : public NullMusicPlugin {
-public:
-	const char *getName() const {
-		return _s("C64 Audio Emulator");
-	}
-
-	const char *getId() const {
-		return "C64";
-	}
-
-	MusicDevices getDevices() const;
-};
-
-MusicDevices C64MusicPlugin::getDevices() const {
-	MusicDevices devices;
-	devices.push_back(MusicDevice(this, "", MT_C64));
-	return devices;
-}
-
-//#if PLUGIN_ENABLED_DYNAMIC(C64)
-	//REGISTER_PLUGIN_DYNAMIC(C64, PLUGIN_TYPE_MUSIC, C64MusicPlugin);
-//#else
-	REGISTER_PLUGIN_STATIC(C64, PLUGIN_TYPE_MUSIC, C64MusicPlugin);
-//#endif
-
-#endif
diff --git a/sound/softsynth/sid.h b/sound/softsynth/sid.h
deleted file mode 100644
index c78f538441..0000000000
--- a/sound/softsynth/sid.h
+++ /dev/null
@@ -1,348 +0,0 @@
-/* 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.
- *
- * $URL$
- * $Id$
- *
- */
-
-/*
- *  This file is based on reSID, a MOS6581 SID emulator engine.
- *  Copyright (C) 2004  Dag Lem <resid@nimrod.no>
- */
-
-#ifndef __SID_H__
-#define __SID_H__
-
-// Inlining on/off.
-#define RESID_INLINE inline
-
-namespace Resid {
-
-// We could have used the smallest possible data type for each SID register,
-// however this would give a slower engine because of data type conversions.
-// An int is assumed to be at least 32 bits (necessary in the types reg24,
-// cycle_count, and sound_sample). GNU does not support 16-bit machines
-// (GNU Coding Standards: Portability between CPUs), so this should be
-// a valid assumption.
-
-typedef unsigned int reg4;
-typedef unsigned int reg8;
-typedef unsigned int reg12;
-typedef unsigned int reg16;
-typedef unsigned int reg24;
-
-typedef int cycle_count;
-typedef int sound_sample;
-typedef sound_sample fc_point[2];
-
-
-class WaveformGenerator {
-public:
-	WaveformGenerator();
-
-	void set_sync_source(WaveformGenerator*);
-
-	void clock(cycle_count delta_t);
-	void synchronize();
-	void reset();
-
-	void writeFREQ_LO(reg8);
-	void writeFREQ_HI(reg8);
-	void writePW_LO(reg8);
-	void writePW_HI(reg8);
-	void writeCONTROL_REG(reg8);
-	reg8 readOSC();
-
-	// 12-bit waveform output.
-	reg12 output();
-
-protected:
-	const WaveformGenerator* sync_source;
-	WaveformGenerator* sync_dest;
-
-	// Tell whether the accumulator MSB was set high on this cycle.
-	bool msb_rising;
-
-	reg24 accumulator;
-	reg24 shift_register;
-
-	// Fout  = (Fn*Fclk/16777216)Hz
-	reg16 freq;
-	// PWout = (PWn/40.95)%
-	reg12 pw;
-
-	// The control register right-shifted 4 bits; used for output function
-	// table lookup.
-	reg8 waveform;
-
-	// The remaining control register bits.
-	reg8 test;
-	reg8 ring_mod;
-	reg8 sync;
-	// The gate bit is handled by the EnvelopeGenerator.
-
-	// 16 possible combinations of waveforms.
-	reg12 output____();
-	reg12 output___T();
-	reg12 output__S_();
-	reg12 output__ST();
-	reg12 output_P__();
-	reg12 output_P_T();
-	reg12 output_PS_();
-	reg12 output_PST();
-	reg12 outputN___();
-	reg12 outputN__T();
-	reg12 outputN_S_();
-	reg12 outputN_ST();
-	reg12 outputNP__();
-	reg12 outputNP_T();
-	reg12 outputNPS_();
-	reg12 outputNPST();
-
-	// Sample data for combinations of waveforms.
-	static const reg8 wave6581__ST[];
-	static const reg8 wave6581_P_T[];
-	static const reg8 wave6581_PS_[];
-	static const reg8 wave6581_PST[];
-
-	friend class Voice;
-	friend class SID;
-};
-
-class Filter {
-public:
-	Filter();
-
-	void enable_filter(bool enable);
-
-	void clock(cycle_count delta_t,
-		sound_sample voice1, sound_sample voice2, sound_sample voice3);
-	void reset();
-
-	// Write registers.
-	void writeFC_LO(reg8);
-	void writeFC_HI(reg8);
-	void writeRES_FILT(reg8);
-	void writeMODE_VOL(reg8);
-
-	// SID audio output (16 bits).
-	sound_sample output();
-
-protected:
-	void set_w0();
-	void set_Q();
-
-	// Filter enabled.
-	bool enabled;
-
-	// Filter cutoff frequency.
-	reg12 fc;
-
-	// Filter resonance.
-	reg8 res;
-
-	// Selects which inputs to route through filter.
-	reg8 filt;
-
-	// Switch voice 3 off.
-	reg8 voice3off;
-
-	// Highpass, bandpass, and lowpass filter modes.
-	reg8 hp_bp_lp;
-
-	// Output master volume.
-	reg4 vol;
-
-	// Mixer DC offset.
-	sound_sample mixer_DC;
-
-	// State of filter.
-	sound_sample Vhp; // highpass
-	sound_sample Vbp; // bandpass
-	sound_sample Vlp; // lowpass
-	sound_sample Vnf; // not filtered
-
-	// Cutoff frequency, resonance.
-	sound_sample w0, w0_ceil_1, w0_ceil_dt;
-	sound_sample _1024_div_Q;
-
-	// Cutoff frequency tables.
-	// FC is an 11 bit register.
-	sound_sample f0_6581[2048];
-	sound_sample* f0;
-	static fc_point f0_points_6581[];
-	fc_point* f0_points;
-	int f0_count;
-
-	friend class SID;
-};
-
-class EnvelopeGenerator {
-public:
-	EnvelopeGenerator();
-
-	enum State { ATTACK, DECAY_SUSTAIN, RELEASE };
-
-	void clock(cycle_count delta_t);
-	void reset();
-
-	void writeCONTROL_REG(reg8);
-	void writeATTACK_DECAY(reg8);
-	void writeSUSTAIN_RELEASE(reg8);
-	reg8 readENV();
-
-	// 8-bit envelope output.
-	reg8 output();
-
-protected:
-	reg16 rate_counter;
-	reg16 rate_period;
-	reg8 exponential_counter;
-	reg8 exponential_counter_period;
-	reg8 envelope_counter;
-	bool hold_zero;
-
-	reg4 attack;
-	reg4 decay;
-	reg4 sustain;
-	reg4 release;
-
-	reg8 gate;
-
-	State state;
-
-	// Lookup table to convert from attack, decay, or release value to rate
-	// counter period.
-	static reg16 rate_counter_period[];
-
-	// The 16 selectable sustain levels.
-	static reg8 sustain_level[];
-
-	friend class SID;
-};
-
-class ExternalFilter {
-public:
-	ExternalFilter();
-
-	void enable_filter(bool enable);
-	void set_sampling_parameter(double pass_freq);
-
-	void clock(cycle_count delta_t, sound_sample Vi);
-	void reset();
-
-	// Audio output (20 bits).
-	sound_sample output();
-
-protected:
-	// Filter enabled.
-	bool enabled;
-
-	// Maximum mixer DC offset.
-	sound_sample mixer_DC;
-
-	// State of filters.
-	sound_sample Vlp; // lowpass
-	sound_sample Vhp; // highpass
-	sound_sample Vo;
-
-	// Cutoff frequencies.
-	sound_sample w0lp;
-	sound_sample w0hp;
-
-	friend class SID;
-};
-
-class Voice {
-public:
-	Voice();
-
-	void set_sync_source(Voice*);
-	void reset();
-
-	void writeCONTROL_REG(reg8);
-
-	// Amplitude modulated waveform output.
-	// Range [-2048*255, 2047*255].
-	sound_sample output() {
-		// Multiply oscillator output with envelope output.
-		return (wave.output() - wave_zero)*envelope.output() + voice_DC;
-	}
-
-protected:
-	WaveformGenerator wave;
-	EnvelopeGenerator envelope;
-
-	// Waveform D/A zero level.
-	sound_sample wave_zero;
-
-	// Multiplying D/A DC offset.
-	sound_sample voice_DC;
-
-	friend class SID;
-};
-
-
-class SID {
-public:
-	SID();
-	~SID();
-
-	void enable_filter(bool enable);
-	void enable_external_filter(bool enable);
-	bool set_sampling_parameters(double clock_freq,
-		double sample_freq, double pass_freq = -1,
-		double filter_scale = 0.97);
-
-	void clock(cycle_count delta_t);
-	int clock(cycle_count& delta_t, short* buf, int n, int interleave = 1);
-	void reset();
-
-	// Read/write registers.
-	reg8 read(reg8 offset);
-	void write(reg8 offset, reg8 value);
-
-	// 16-bit output (AUDIO OUT).
-	int output();
-
-protected:
-	Voice voice[3];
-	Filter filter;
-	ExternalFilter extfilt;
-
-	reg8 bus_value;
-	cycle_count bus_value_ttl;
-
-	double clock_frequency;
-
-	// Fixpoint constants.
-	static const int FIXP_SHIFT;
-	static const int FIXP_MASK;
-
-	// Sampling variables.
-	cycle_count cycles_per_sample;
-	cycle_count sample_offset;
-	short sample_prev;
-};
-
-}
-
-#endif // not __SID_H__
diff --git a/sound/softsynth/wave6581.cpp b/sound/softsynth/wave6581.cpp
deleted file mode 100644
index d1ddad1623..0000000000
--- a/sound/softsynth/wave6581.cpp
+++ /dev/null
@@ -1,2098 +0,0 @@
-/* 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.
- *
- * $URL$
- * $Id$
- *
- */
-
-/*
- *  This file is based on reSID, a MOS6581 SID emulator engine.
- *  Copyright (C) 2004  Dag Lem <resid@nimrod.no>
- */
-
-#ifndef DISABLE_SID
-
-#include "sid.h"
-
-namespace Resid {
-
-const reg8 WaveformGenerator::wave6581__ST[] = {
-/* 0x000: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x008: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x010: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x018: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x020: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x028: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x030: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x038: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x040: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x048: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x050: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x058: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x060: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x068: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x070: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x078: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x03, 0x03,
-/* 0x080: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x088: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x090: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x098: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
-/* 0x0c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0e8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0f0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0f8: */  0x00, 0x00, 0x00, 0x00, 0x07, 0x07, 0x07, 0x07,
-/* 0x100: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x108: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x110: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x118: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x120: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x128: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x130: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x138: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
-/* 0x140: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x148: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x150: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x158: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x160: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x168: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x170: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x178: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x03, 0x03,
-/* 0x180: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x188: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x190: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x198: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
-/* 0x1c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1e8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1f0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1f8: */  0x0e, 0x0e, 0x0e, 0x0e, 0x0f, 0x0f, 0x0f, 0x0f,
-/* 0x200: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x208: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x210: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x218: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x220: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x228: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x230: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x238: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x240: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x248: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x250: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x258: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x260: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x268: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x270: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x278: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x03, 0x03,
-/* 0x280: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x288: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x290: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x298: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
-/* 0x2c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2e8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2f0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2f8: */  0x00, 0x00, 0x00, 0x00, 0x07, 0x07, 0x07, 0x07,
-/* 0x300: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x308: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x310: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x318: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x320: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x328: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x330: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x338: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
-/* 0x340: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x348: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x350: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x358: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x360: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x368: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x370: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x378: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x03, 0x03,
-/* 0x380: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x388: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x390: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x398: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x3a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x3a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x3b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x3b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
-/* 0x3c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x3c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x3d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x3d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x3e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x3e8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x3f0: */  0x1c, 0x1c, 0x1c, 0x1c, 0x1c, 0x1c, 0x1c, 0x1c,
-/* 0x3f8: */  0x1e, 0x1e, 0x1e, 0x1e, 0x1f, 0x1f, 0x3f, 0x3f,
-/* 0x400: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x408: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x410: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x418: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x420: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x428: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x430: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x438: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x440: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x448: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x450: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x458: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x460: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x468: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x470: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x478: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x03, 0x03,
-/* 0x480: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x488: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x490: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x498: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
-/* 0x4c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4e8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4f0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4f8: */  0x00, 0x00, 0x00, 0x00, 0x07, 0x07, 0x07, 0x07,
-/* 0x500: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x508: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x510: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x518: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x520: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x528: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x530: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x538: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
-/* 0x540: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x548: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x550: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x558: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x560: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x568: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x570: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x578: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x03, 0x03,
-/* 0x580: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x588: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x590: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x598: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x5a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x5a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x5b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x5b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
-/* 0x5c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x5c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x5d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x5d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x5e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x5e8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x5f0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x5f8: */  0x0e, 0x0e, 0x0e, 0x0e, 0x0f, 0x0f, 0x0f, 0x1f,
-/* 0x600: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x608: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x610: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x618: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x620: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x628: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x630: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x638: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x640: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x648: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x650: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x658: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x660: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x668: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x670: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x678: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x03, 0x03,
-/* 0x680: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x688: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x690: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x698: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x6a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x6a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x6b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x6b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
-/* 0x6c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x6c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x6d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x6d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x6e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x6e8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x6f0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x6f8: */  0x00, 0x00, 0x00, 0x00, 0x07, 0x07, 0x07, 0x07,
-/* 0x700: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x708: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x710: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x718: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x720: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x728: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x730: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x738: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
-/* 0x740: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x748: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x750: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x758: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x760: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x768: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x770: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x778: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x03, 0x03,
-/* 0x780: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x788: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x790: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x798: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x7a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x7a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x7b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x7b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
-/* 0x7c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x7c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x7d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x7d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x7e0: */  0x38, 0x38, 0x38, 0x38, 0x38, 0x38, 0x38, 0x38,
-/* 0x7e8: */  0x38, 0x38, 0x38, 0x38, 0x38, 0x38, 0x38, 0x38,
-/* 0x7f0: */  0x3c, 0x3c, 0x3c, 0x3c, 0x3c, 0x3c, 0x3c, 0x3c,
-/* 0x7f8: */  0x3e, 0x3e, 0x3f, 0x3f, 0x7f, 0x7f, 0x7f, 0x7f,
-/* 0x800: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x808: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x810: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x818: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x820: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x828: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x830: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x838: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x840: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x848: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x850: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x858: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x860: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x868: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x870: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x878: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x03, 0x03,
-/* 0x880: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x888: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x890: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x898: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
-/* 0x8c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8e8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8f0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8f8: */  0x00, 0x00, 0x00, 0x00, 0x07, 0x07, 0x07, 0x07,
-/* 0x900: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x908: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x910: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x918: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x920: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x928: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x930: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x938: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
-/* 0x940: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x948: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x950: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x958: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x960: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x968: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x970: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x978: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x03, 0x03,
-/* 0x980: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x988: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x990: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x998: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
-/* 0x9c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9e8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9f0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9f8: */  0x0e, 0x0e, 0x0e, 0x0e, 0x0f, 0x0f, 0x0f, 0x0f,
-/* 0xa00: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa08: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa10: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa18: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa20: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa28: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa30: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa38: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa40: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa48: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa50: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa58: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa60: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa68: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa70: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa78: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x03, 0x03,
-/* 0xa80: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa88: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa90: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa98: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xaa0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xaa8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xab0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xab8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
-/* 0xac0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xac8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xad0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xad8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xae0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xae8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xaf0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xaf8: */  0x00, 0x00, 0x00, 0x00, 0x07, 0x07, 0x07, 0x07,
-/* 0xb00: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb08: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb10: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb18: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb20: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb28: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb30: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb38: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
-/* 0xb40: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb48: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb50: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb58: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb60: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb68: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb70: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb78: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x03, 0x03,
-/* 0xb80: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb88: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb90: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb98: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xba0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xba8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbb0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbb8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
-/* 0xbc0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbc8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbd0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbd8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbe0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbe8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbf0: */  0x1c, 0x1c, 0x1c, 0x1c, 0x1c, 0x1c, 0x1c, 0x1c,
-/* 0xbf8: */  0x1e, 0x1e, 0x1e, 0x1e, 0x1f, 0x1f, 0x3f, 0x3f,
-/* 0xc00: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc08: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc10: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc18: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc20: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc28: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc30: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc38: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc40: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc48: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc50: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc58: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc60: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc68: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc70: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc78: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x03, 0x03,
-/* 0xc80: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc88: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc90: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc98: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xca0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xca8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xcb0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xcb8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
-/* 0xcc0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xcc8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xcd0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xcd8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xce0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xce8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xcf0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xcf8: */  0x00, 0x00, 0x00, 0x00, 0x07, 0x07, 0x07, 0x07,
-/* 0xd00: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd08: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd10: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd18: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd20: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd28: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd30: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd38: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
-/* 0xd40: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd48: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd50: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd58: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd60: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd68: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd70: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd78: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x03, 0x03,
-/* 0xd80: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd88: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd90: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd98: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xda0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xda8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xdb0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xdb8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
-/* 0xdc0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xdc8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xdd0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xdd8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xde0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xde8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xdf0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xdf8: */  0x0e, 0x0e, 0x0e, 0x0e, 0x0f, 0x0f, 0x0f, 0x1f,
-/* 0xe00: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe08: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe10: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe18: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe20: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe28: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe30: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe38: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe40: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe48: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe50: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe58: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe60: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe68: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe70: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe78: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x03, 0x03,
-/* 0xe80: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe88: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe90: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe98: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xea0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xea8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xeb0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xeb8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
-/* 0xec0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xec8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xed0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xed8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xee0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xee8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xef0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xef8: */  0x00, 0x00, 0x00, 0x00, 0x07, 0x07, 0x07, 0x07,
-/* 0xf00: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf08: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf10: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf18: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf20: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf28: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf30: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf38: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
-/* 0xf40: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf48: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf50: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf58: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf60: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf68: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf70: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf78: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x03, 0x03,
-/* 0xf80: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf88: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf90: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf98: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xfa0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xfa8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xfb0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xfb8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
-/* 0xfc0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xfc8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xfd0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xfd8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xfe0: */  0x38, 0x38, 0x38, 0x38, 0x38, 0x38, 0x38, 0x38,
-/* 0xfe8: */  0x38, 0x38, 0x38, 0x38, 0x38, 0x38, 0x38, 0x38,
-/* 0xff0: */  0x3c, 0x3c, 0x3c, 0x3c, 0x3c, 0x3c, 0x3c, 0x3c,
-/* 0xff8: */  0x3e, 0x3e, 0x3f, 0x3f, 0x7f, 0x7f, 0x7f, 0x7f,
-};
-
-const reg8 WaveformGenerator::wave6581_P_T[] = {
-/* 0x000: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x008: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x010: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x018: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x020: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x028: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x030: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x038: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x040: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x048: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x050: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x058: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x060: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x068: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x070: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x078: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x080: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x088: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x090: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x098: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0e8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0f0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0f8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x100: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x108: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x110: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x118: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x120: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x128: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x130: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x138: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x140: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x148: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x150: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x158: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x160: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x168: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x170: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x178: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x180: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x188: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x190: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x198: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1e8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1f0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1f8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x20, 0x38, 0x3f,
-/* 0x200: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x208: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x210: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x218: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x220: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x228: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x230: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x238: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x240: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x248: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x250: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x258: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x260: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x268: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x270: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x278: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x280: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x288: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x290: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x298: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2e8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2f0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2f8: */  0x00, 0x00, 0x00, 0x40, 0x00, 0x40, 0x40, 0x5f,
-/* 0x300: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x308: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x310: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x318: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x320: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x328: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x330: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x338: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x340: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x348: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x350: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x358: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x360: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x368: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x40,
-/* 0x370: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x40,
-/* 0x378: */  0x00, 0x00, 0x00, 0x60, 0x00, 0x60, 0x60, 0x6f,
-/* 0x380: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x388: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x390: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x398: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x40,
-/* 0x3a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x3a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x60,
-/* 0x3b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x60,
-/* 0x3b8: */  0x00, 0x00, 0x00, 0x60, 0x00, 0x60, 0x70, 0x77,
-/* 0x3c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x3c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x60,
-/* 0x3d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x60,
-/* 0x3d8: */  0x00, 0x00, 0x00, 0x70, 0x40, 0x70, 0x70, 0x7b,
-/* 0x3e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x40, 0x70,
-/* 0x3e8: */  0x00, 0x40, 0x40, 0x70, 0x60, 0x70, 0x78, 0x7d,
-/* 0x3f0: */  0x00, 0x40, 0x60, 0x78, 0x60, 0x78, 0x78, 0x7e,
-/* 0x3f8: */  0x70, 0x7c, 0x7c, 0x7f, 0x7e, 0x7f, 0x7f, 0x7f,
-/* 0x400: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x408: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x410: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x418: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x420: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x428: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x430: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x438: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x440: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x448: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x450: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x458: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x460: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x468: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x470: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x478: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x80,
-/* 0x480: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x488: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x490: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x498: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x80,
-/* 0x4c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x80,
-/* 0x4e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4e8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x80,
-/* 0x4f0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x80,
-/* 0x4f8: */  0x00, 0x00, 0x00, 0x80, 0x80, 0x80, 0x80, 0x9f,
-/* 0x500: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x508: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x510: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x518: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x520: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x528: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x530: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x538: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x80,
-/* 0x540: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x548: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x550: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x558: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x80,
-/* 0x560: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x568: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x80,
-/* 0x570: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x80,
-/* 0x578: */  0x00, 0x80, 0x80, 0x80, 0x80, 0xa0, 0xa0, 0xaf,
-/* 0x580: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x588: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x590: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x598: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x80,
-/* 0x5a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x5a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x80, 0x80, 0x80,
-/* 0x5b0: */  0x00, 0x00, 0x00, 0x80, 0x00, 0x80, 0x80, 0xa0,
-/* 0x5b8: */  0x00, 0x80, 0x80, 0xa0, 0x80, 0xa0, 0xb0, 0xb7,
-/* 0x5c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x80,
-/* 0x5c8: */  0x00, 0x00, 0x00, 0x80, 0x00, 0x80, 0x80, 0xa0,
-/* 0x5d0: */  0x00, 0x00, 0x00, 0x80, 0x00, 0x80, 0x80, 0xa0,
-/* 0x5d8: */  0x00, 0x80, 0x80, 0xa0, 0x80, 0xb0, 0xb0, 0xbb,
-/* 0x5e0: */  0x00, 0x00, 0x00, 0x80, 0x80, 0x80, 0x80, 0xb0,
-/* 0x5e8: */  0x80, 0x80, 0x80, 0xb0, 0x80, 0xb0, 0xb8, 0xbd,
-/* 0x5f0: */  0x80, 0x80, 0x80, 0xb8, 0xa0, 0xb8, 0xb8, 0xbe,
-/* 0x5f8: */  0xa0, 0xb8, 0xbc, 0xbf, 0xbe, 0xbf, 0xbf, 0xbf,
-/* 0x600: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x608: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x610: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x618: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x620: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x628: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x630: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x638: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xc0,
-/* 0x640: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x648: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x650: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x658: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x80, 0x80, 0xc0,
-/* 0x660: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x668: */  0x00, 0x00, 0x00, 0x80, 0x00, 0x80, 0x80, 0xc0,
-/* 0x670: */  0x00, 0x00, 0x00, 0x80, 0x00, 0x80, 0x80, 0xc0,
-/* 0x678: */  0x00, 0x80, 0x80, 0xc0, 0xc0, 0xc0, 0xc0, 0xcf,
-/* 0x680: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x688: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x80,
-/* 0x690: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x80,
-/* 0x698: */  0x00, 0x00, 0x00, 0x80, 0x00, 0x80, 0x80, 0xc0,
-/* 0x6a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x80,
-/* 0x6a8: */  0x00, 0x00, 0x00, 0x80, 0x00, 0x80, 0x80, 0xc0,
-/* 0x6b0: */  0x00, 0x00, 0x00, 0x80, 0x00, 0x80, 0xc0, 0xc0,
-/* 0x6b8: */  0x80, 0xc0, 0xc0, 0xc0, 0xc0, 0xc0, 0xd0, 0xd7,
-/* 0x6c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x80,
-/* 0x6c8: */  0x00, 0x00, 0x00, 0x80, 0x80, 0x80, 0xc0, 0xc0,
-/* 0x6d0: */  0x00, 0x80, 0x80, 0xc0, 0x80, 0xc0, 0xc0, 0xc0,
-/* 0x6d8: */  0x80, 0xc0, 0xc0, 0xc0, 0xc0, 0xd0, 0xd0, 0xdb,
-/* 0x6e0: */  0x00, 0x80, 0x80, 0xc0, 0x80, 0xc0, 0xc0, 0xd0,
-/* 0x6e8: */  0x80, 0xc0, 0xc0, 0xd0, 0xc0, 0xd0, 0xd8, 0xdd,
-/* 0x6f0: */  0xc0, 0xc0, 0xc0, 0xd0, 0xc0, 0xd8, 0xd8, 0xde,
-/* 0x6f8: */  0xc0, 0xd8, 0xdc, 0xdf, 0xdc, 0xdf, 0xdf, 0xdf,
-/* 0x700: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x708: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x80,
-/* 0x710: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x80,
-/* 0x718: */  0x00, 0x00, 0x00, 0x80, 0x80, 0xc0, 0xc0, 0xe0,
-/* 0x720: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x80,
-/* 0x728: */  0x00, 0x80, 0x80, 0xc0, 0x80, 0xc0, 0xc0, 0xe0,
-/* 0x730: */  0x00, 0x80, 0x80, 0xc0, 0x80, 0xc0, 0xc0, 0xe0,
-/* 0x738: */  0x80, 0xc0, 0xc0, 0xe0, 0xc0, 0xe0, 0xe0, 0xe7,
-/* 0x740: */  0x00, 0x00, 0x00, 0x80, 0x00, 0x80, 0x80, 0xc0,
-/* 0x748: */  0x00, 0x80, 0x80, 0xc0, 0x80, 0xc0, 0xc0, 0xe0,
-/* 0x750: */  0x00, 0x80, 0x80, 0xc0, 0x80, 0xc0, 0xc0, 0xe0,
-/* 0x758: */  0xc0, 0xc0, 0xc0, 0xe0, 0xe0, 0xe0, 0xe0, 0xeb,
-/* 0x760: */  0x80, 0x80, 0xc0, 0xc0, 0xc0, 0xc0, 0xc0, 0xe0,
-/* 0x768: */  0xc0, 0xe0, 0xe0, 0xe0, 0xe0, 0xe0, 0xe0, 0xed,
-/* 0x770: */  0xc0, 0xe0, 0xe0, 0xe0, 0xe0, 0xe8, 0xe8, 0xee,
-/* 0x778: */  0xe0, 0xe8, 0xec, 0xef, 0xec, 0xef, 0xef, 0xef,
-/* 0x780: */  0x00, 0x00, 0x00, 0x80, 0x80, 0x80, 0x80, 0xc0,
-/* 0x788: */  0x80, 0x80, 0xc0, 0xc0, 0xc0, 0xc0, 0xc0, 0xf0,
-/* 0x790: */  0x80, 0xc0, 0xc0, 0xc0, 0xc0, 0xe0, 0xe0, 0xf0,
-/* 0x798: */  0xc0, 0xe0, 0xe0, 0xf0, 0xe0, 0xf0, 0xf0, 0xf3,
-/* 0x7a0: */  0x80, 0xc0, 0xc0, 0xe0, 0xc0, 0xe0, 0xe0, 0xf0,
-/* 0x7a8: */  0xc0, 0xe0, 0xe0, 0xf0, 0xe0, 0xf0, 0xf0, 0xf5,
-/* 0x7b0: */  0xe0, 0xe0, 0xe0, 0xf0, 0xf0, 0xf0, 0xf0, 0xf6,
-/* 0x7b8: */  0xf0, 0xf0, 0xf4, 0xf7, 0xf4, 0xf7, 0xf7, 0xf7,
-/* 0x7c0: */  0xc0, 0xc0, 0xc0, 0xe0, 0xe0, 0xe0, 0xe0, 0xf0,
-/* 0x7c8: */  0xe0, 0xe0, 0xe0, 0xf8, 0xf0, 0xf8, 0xf8, 0xf9,
-/* 0x7d0: */  0xe0, 0xf0, 0xf0, 0xf8, 0xf0, 0xf8, 0xf8, 0xfa,
-/* 0x7d8: */  0xf0, 0xf8, 0xf8, 0xfb, 0xf8, 0xfb, 0xfb, 0xfb,
-/* 0x7e0: */  0xe0, 0xf0, 0xf0, 0xf8, 0xf0, 0xf8, 0xfc, 0xfc,
-/* 0x7e8: */  0xf8, 0xfc, 0xfc, 0xfd, 0xfc, 0xfd, 0xfd, 0xfd,
-/* 0x7f0: */  0xf8, 0xfc, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe,
-/* 0x7f8: */  0xfe, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
-/* 0x800: */  0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xfe,
-/* 0x808: */  0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfc, 0xf8,
-/* 0x810: */  0xfd, 0xfd, 0xfd, 0xfc, 0xfd, 0xfc, 0xfc, 0xf8,
-/* 0x818: */  0xfc, 0xfc, 0xfc, 0xf0, 0xf8, 0xf0, 0xf0, 0xe0,
-/* 0x820: */  0xfb, 0xfb, 0xfb, 0xf8, 0xfb, 0xf8, 0xf8, 0xf0,
-/* 0x828: */  0xfa, 0xf8, 0xf8, 0xf0, 0xf8, 0xf0, 0xf0, 0xe0,
-/* 0x830: */  0xf9, 0xf8, 0xf8, 0xf0, 0xf8, 0xf0, 0xe0, 0xe0,
-/* 0x838: */  0xf0, 0xe0, 0xe0, 0xe0, 0xe0, 0xc0, 0xc0, 0xc0,
-/* 0x840: */  0xf7, 0xf7, 0xf7, 0xf4, 0xf7, 0xf4, 0xf0, 0xf0,
-/* 0x848: */  0xf6, 0xf0, 0xf0, 0xf0, 0xf0, 0xe0, 0xe0, 0xe0,
-/* 0x850: */  0xf5, 0xf0, 0xf0, 0xe0, 0xf0, 0xe0, 0xe0, 0xc0,
-/* 0x858: */  0xf0, 0xe0, 0xe0, 0xc0, 0xe0, 0xc0, 0xc0, 0x80,
-/* 0x860: */  0xf3, 0xf0, 0xf0, 0xe0, 0xf0, 0xe0, 0xe0, 0xc0,
-/* 0x868: */  0xf0, 0xe0, 0xe0, 0xc0, 0xc0, 0xc0, 0xc0, 0x80,
-/* 0x870: */  0xf0, 0xe0, 0xc0, 0xc0, 0xc0, 0xc0, 0x80, 0x80,
-/* 0x878: */  0xc0, 0x80, 0x80, 0x80, 0x80, 0x00, 0x00, 0x00,
-/* 0x880: */  0xef, 0xef, 0xef, 0xec, 0xef, 0xec, 0xe8, 0xe0,
-/* 0x888: */  0xee, 0xe8, 0xe8, 0xe0, 0xe0, 0xe0, 0xe0, 0xc0,
-/* 0x890: */  0xed, 0xe8, 0xe0, 0xe0, 0xe0, 0xe0, 0xe0, 0xc0,
-/* 0x898: */  0xe0, 0xe0, 0xc0, 0xc0, 0xc0, 0xc0, 0x80, 0x80,
-/* 0x8a0: */  0xeb, 0xe0, 0xe0, 0xe0, 0xe0, 0xc0, 0xc0, 0xc0,
-/* 0x8a8: */  0xe0, 0xc0, 0xc0, 0x80, 0xc0, 0x80, 0x80, 0x00,
-/* 0x8b0: */  0xe0, 0xc0, 0xc0, 0x80, 0xc0, 0x80, 0x80, 0x00,
-/* 0x8b8: */  0xc0, 0x80, 0x80, 0x00, 0x80, 0x00, 0x00, 0x00,
-/* 0x8c0: */  0xe7, 0xe0, 0xe0, 0xc0, 0xe0, 0xc0, 0xc0, 0x80,
-/* 0x8c8: */  0xe0, 0xc0, 0xc0, 0x80, 0xc0, 0x80, 0x80, 0x00,
-/* 0x8d0: */  0xe0, 0xc0, 0xc0, 0x80, 0xc0, 0x80, 0x80, 0x00,
-/* 0x8d8: */  0x80, 0x80, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8e0: */  0xe0, 0xc0, 0xc0, 0x80, 0x80, 0x00, 0x00, 0x00,
-/* 0x8e8: */  0x80, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8f0: */  0x80, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8f8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x900: */  0xdf, 0xdf, 0xdf, 0xdc, 0xdf, 0xdc, 0xd8, 0xc0,
-/* 0x908: */  0xde, 0xd8, 0xd8, 0xc0, 0xd8, 0xc0, 0xc0, 0xc0,
-/* 0x910: */  0xdd, 0xd8, 0xd0, 0xc0, 0xd0, 0xc0, 0xc0, 0x80,
-/* 0x918: */  0xd0, 0xc0, 0xc0, 0x80, 0xc0, 0x80, 0x80, 0x00,
-/* 0x920: */  0xdb, 0xd0, 0xd0, 0xc0, 0xc0, 0xc0, 0xc0, 0x80,
-/* 0x928: */  0xc0, 0xc0, 0xc0, 0x80, 0xc0, 0x80, 0x80, 0x00,
-/* 0x930: */  0xc0, 0xc0, 0x80, 0x80, 0x80, 0x00, 0x00, 0x00,
-/* 0x938: */  0x80, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x940: */  0xd7, 0xd0, 0xc0, 0xc0, 0xc0, 0xc0, 0xc0, 0x80,
-/* 0x948: */  0xc0, 0xc0, 0x80, 0x00, 0x80, 0x00, 0x00, 0x00,
-/* 0x950: */  0xc0, 0x80, 0x80, 0x00, 0x80, 0x00, 0x00, 0x00,
-/* 0x958: */  0x80, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x960: */  0xc0, 0x80, 0x80, 0x00, 0x80, 0x00, 0x00, 0x00,
-/* 0x968: */  0x80, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x970: */  0x80, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x978: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x980: */  0xcf, 0xc0, 0xc0, 0xc0, 0xc0, 0x80, 0x80, 0x00,
-/* 0x988: */  0xc0, 0x80, 0x80, 0x00, 0x80, 0x00, 0x00, 0x00,
-/* 0x990: */  0xc0, 0x80, 0x80, 0x00, 0x80, 0x00, 0x00, 0x00,
-/* 0x998: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9a0: */  0xc0, 0x80, 0x80, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9c0: */  0xc0, 0x80, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9e8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9f0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9f8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa00: */  0xbf, 0xbf, 0xbf, 0xbe, 0xbf, 0xbc, 0xbc, 0xa0,
-/* 0xa08: */  0xbe, 0xbc, 0xb8, 0xa0, 0xb8, 0xa0, 0x80, 0x80,
-/* 0xa10: */  0xbd, 0xb8, 0xb0, 0x80, 0xb0, 0x80, 0x80, 0x80,
-/* 0xa18: */  0xb0, 0x80, 0x80, 0x80, 0x80, 0x00, 0x00, 0x00,
-/* 0xa20: */  0xbb, 0xb0, 0xb0, 0x80, 0xa0, 0x80, 0x80, 0x00,
-/* 0xa28: */  0xa0, 0x80, 0x80, 0x00, 0x80, 0x00, 0x00, 0x00,
-/* 0xa30: */  0xa0, 0x80, 0x80, 0x00, 0x80, 0x00, 0x00, 0x00,
-/* 0xa38: */  0x80, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa40: */  0xb7, 0xb0, 0xa0, 0x80, 0xa0, 0x80, 0x80, 0x00,
-/* 0xa48: */  0xa0, 0x80, 0x80, 0x00, 0x80, 0x00, 0x00, 0x00,
-/* 0xa50: */  0x80, 0x80, 0x80, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa58: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa60: */  0x80, 0x80, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa68: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa70: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa78: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa80: */  0xaf, 0xa0, 0xa0, 0x80, 0x80, 0x80, 0x80, 0x00,
-/* 0xa88: */  0x80, 0x80, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa90: */  0x80, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa98: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xaa0: */  0x80, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xaa8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xab0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xab8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xac0: */  0x80, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xac8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xad0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xad8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xae0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xae8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xaf0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xaf8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb00: */  0x9f, 0x90, 0x80, 0x80, 0x80, 0x00, 0x00, 0x00,
-/* 0xb08: */  0x80, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb10: */  0x80, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb18: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb20: */  0x80, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb28: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb30: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb38: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb40: */  0x80, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb48: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb50: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb58: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb60: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb68: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb70: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb78: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb80: */  0x80, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb88: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb90: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb98: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xba0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xba8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbb0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbb8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbc0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbc8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbd0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbd8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbe0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbe8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbf0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbf8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc00: */  0x7f, 0x7f, 0x7f, 0x7e, 0x7f, 0x7c, 0x7c, 0x70,
-/* 0xc08: */  0x7e, 0x7c, 0x78, 0x60, 0x78, 0x60, 0x60, 0x00,
-/* 0xc10: */  0x7d, 0x78, 0x78, 0x60, 0x70, 0x40, 0x40, 0x00,
-/* 0xc18: */  0x70, 0x40, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc20: */  0x7b, 0x78, 0x70, 0x40, 0x70, 0x40, 0x00, 0x00,
-/* 0xc28: */  0x60, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc30: */  0x60, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc38: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc40: */  0x77, 0x70, 0x70, 0x00, 0x60, 0x00, 0x00, 0x00,
-/* 0xc48: */  0x60, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc50: */  0x60, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc58: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc60: */  0x40, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc68: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc70: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc78: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc80: */  0x6f, 0x60, 0x60, 0x00, 0x60, 0x00, 0x00, 0x00,
-/* 0xc88: */  0x40, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc90: */  0x40, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc98: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xca0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xca8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xcb0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xcb8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xcc0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xcc8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xcd0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xcd8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xce0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xce8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xcf0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xcf8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd00: */  0x5f, 0x58, 0x40, 0x00, 0x40, 0x00, 0x00, 0x00,
-/* 0xd08: */  0x40, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd10: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd18: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd20: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd28: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd30: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd38: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd40: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd48: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd50: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd58: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd60: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd68: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd70: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd78: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd80: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd88: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd90: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd98: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xda0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xda8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xdb0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xdb8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xdc0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xdc8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xdd0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xdd8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xde0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xde8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xdf0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xdf8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe00: */  0x3f, 0x3c, 0x30, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe08: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe10: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe18: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe20: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe28: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe30: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe38: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe40: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe48: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe50: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe58: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe60: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe68: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe70: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe78: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe80: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe88: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe90: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe98: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xea0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xea8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xeb0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xeb8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xec0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xec8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xed0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xed8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xee0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xee8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xef0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xef8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf00: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf08: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf10: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf18: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf20: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf28: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf30: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf38: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf40: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf48: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf50: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf58: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf60: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf68: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf70: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf78: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf80: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf88: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf90: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf98: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xfa0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xfa8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xfb0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xfb8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xfc0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xfc8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xfd0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xfd8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xfe0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xfe8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xff0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xff8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-};
-
-const reg8 WaveformGenerator::wave6581_PS_[] = {
-/* 0x000: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x008: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x010: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x018: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x020: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x028: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x030: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x038: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x040: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x048: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x050: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x058: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x060: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x068: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x070: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x078: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x080: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x088: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x090: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x098: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0e8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0f0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0f8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x07,
-/* 0x100: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x108: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x110: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x118: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x120: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x128: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x130: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x138: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x140: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x148: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x150: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x158: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x160: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x168: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x170: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x178: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x03,
-/* 0x180: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x188: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x190: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x198: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x03,
-/* 0x1c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1e8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1f0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1f8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x1f,
-/* 0x200: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x208: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x210: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x218: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x220: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x228: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x230: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x238: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x240: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x248: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x250: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x258: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x260: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x268: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x270: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x278: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x03,
-/* 0x280: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x288: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x290: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x298: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
-/* 0x2c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2e8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2f0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2f8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x2f,
-/* 0x300: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x308: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x310: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x318: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x320: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x328: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x330: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x338: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x340: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x348: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x350: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x358: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x360: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x368: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x370: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x378: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x37,
-/* 0x380: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x388: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x390: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x398: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x3a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x3a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x3b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x3b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x3b,
-/* 0x3c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x3c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x3d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x3d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x3d,
-/* 0x3e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x3e8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x3e,
-/* 0x3f0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x30, 0x3f,
-/* 0x3f8: */  0x00, 0x30, 0x38, 0x3f, 0x3e, 0x3f, 0x3f, 0x3f,
-/* 0x400: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x408: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x410: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x418: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x420: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x428: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x430: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x438: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x440: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x448: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x450: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x458: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x460: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x468: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x470: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x478: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x03,
-/* 0x480: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x488: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x490: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x498: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4e8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4f0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4f8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x4f,
-/* 0x500: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x508: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x510: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x518: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x520: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x528: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x530: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x538: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x540: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x548: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x550: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x558: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x560: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x568: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x570: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x578: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x57,
-/* 0x580: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x588: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x590: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x598: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x5a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x5a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x5b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x5b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x5b,
-/* 0x5c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x5c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x5d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x5d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x5d,
-/* 0x5e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x5e8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x5e,
-/* 0x5f0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x40, 0x5f,
-/* 0x5f8: */  0x00, 0x40, 0x40, 0x5f, 0x5c, 0x5f, 0x5f, 0x5f,
-/* 0x600: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x608: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x610: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x618: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x620: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x628: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x630: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x638: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x640: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x648: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x650: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x658: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x660: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x668: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x670: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x678: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x67,
-/* 0x680: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x688: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x690: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x698: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x6a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x6a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x6b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x6b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x40, 0x6b,
-/* 0x6c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x6c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x6d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x6d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x40, 0x40, 0x6d,
-/* 0x6e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x40,
-/* 0x6e8: */  0x00, 0x00, 0x00, 0x40, 0x00, 0x40, 0x40, 0x6e,
-/* 0x6f0: */  0x00, 0x00, 0x00, 0x40, 0x00, 0x60, 0x60, 0x6f,
-/* 0x6f8: */  0x00, 0x60, 0x60, 0x6f, 0x60, 0x6f, 0x6f, 0x6f,
-/* 0x700: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x708: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x710: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x718: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x40,
-/* 0x720: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x728: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x40,
-/* 0x730: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x40,
-/* 0x738: */  0x00, 0x00, 0x00, 0x40, 0x00, 0x40, 0x60, 0x73,
-/* 0x740: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x748: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x40,
-/* 0x750: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x40,
-/* 0x758: */  0x00, 0x00, 0x00, 0x40, 0x00, 0x60, 0x60, 0x75,
-/* 0x760: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x60,
-/* 0x768: */  0x00, 0x00, 0x00, 0x60, 0x00, 0x60, 0x60, 0x76,
-/* 0x770: */  0x00, 0x00, 0x00, 0x60, 0x00, 0x60, 0x60, 0x77,
-/* 0x778: */  0x00, 0x70, 0x70, 0x77, 0x70, 0x77, 0x77, 0x77,
-/* 0x780: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x788: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x60,
-/* 0x790: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x60,
-/* 0x798: */  0x00, 0x00, 0x00, 0x60, 0x00, 0x60, 0x60, 0x79,
-/* 0x7a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x60,
-/* 0x7a8: */  0x00, 0x00, 0x00, 0x60, 0x00, 0x70, 0x70, 0x7a,
-/* 0x7b0: */  0x00, 0x00, 0x00, 0x70, 0x00, 0x70, 0x70, 0x7b,
-/* 0x7b8: */  0x40, 0x70, 0x70, 0x7b, 0x78, 0x7b, 0x7b, 0x7b,
-/* 0x7c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x70,
-/* 0x7c8: */  0x00, 0x00, 0x00, 0x70, 0x00, 0x70, 0x70, 0x7c,
-/* 0x7d0: */  0x00, 0x00, 0x00, 0x70, 0x40, 0x70, 0x70, 0x7d,
-/* 0x7d8: */  0x40, 0x70, 0x78, 0x7d, 0x78, 0x7d, 0x7d, 0x7d,
-/* 0x7e0: */  0x00, 0x40, 0x40, 0x78, 0x60, 0x78, 0x78, 0x7e,
-/* 0x7e8: */  0x60, 0x78, 0x78, 0x7e, 0x7c, 0x7e, 0x7e, 0x7e,
-/* 0x7f0: */  0x70, 0x7c, 0x7c, 0x7f, 0x7e, 0x7f, 0x7f, 0x7f,
-/* 0x7f8: */  0x7e, 0x7f, 0x7f, 0x7f, 0x7f, 0x7f, 0x7f, 0x7f,
-/* 0x800: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x808: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x810: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x818: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x820: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x828: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x830: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x838: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x840: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x848: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x850: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x858: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x860: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x868: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x870: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x878: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x880: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x888: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x890: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x898: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8e8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8f0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8f8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x07,
-/* 0x900: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x908: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x910: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x918: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x920: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x928: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x930: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x938: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x940: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x948: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x950: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x958: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x960: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x968: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x970: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x978: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x03,
-/* 0x980: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x988: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x990: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x998: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x03,
-/* 0x9c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9e8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9f0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9f8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x1f,
-/* 0xa00: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa08: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa10: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa18: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa20: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa28: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa30: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa38: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa40: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa48: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa50: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa58: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa60: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa68: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa70: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa78: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x03,
-/* 0xa80: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa88: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa90: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa98: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xaa0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xaa8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xab0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xab8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
-/* 0xac0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xac8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xad0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xad8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xae0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xae8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xaf0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xaf8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x2f,
-/* 0xb00: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb08: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb10: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb18: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb20: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb28: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb30: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb38: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb40: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb48: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb50: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb58: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb60: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb68: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb70: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb78: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x37,
-/* 0xb80: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb88: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb90: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb98: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xba0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xba8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbb0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbb8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x3b,
-/* 0xbc0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbc8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbd0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbd8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x3d,
-/* 0xbe0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbe8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x3e,
-/* 0xbf0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x30, 0x3f,
-/* 0xbf8: */  0x00, 0x30, 0x38, 0x3f, 0x3e, 0x3f, 0x3f, 0x3f,
-/* 0xc00: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc08: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc10: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc18: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc20: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc28: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc30: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc38: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc40: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc48: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc50: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc58: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc60: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc68: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc70: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc78: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x03,
-/* 0xc80: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc88: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc90: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc98: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xca0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xca8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xcb0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xcb8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xcc0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xcc8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xcd0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xcd8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xce0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xce8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xcf0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xcf8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x4f,
-/* 0xd00: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd08: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd10: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd18: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd20: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd28: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd30: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd38: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd40: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd48: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd50: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd58: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd60: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd68: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd70: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd78: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x57,
-/* 0xd80: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd88: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd90: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd98: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xda0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xda8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xdb0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xdb8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x5b,
-/* 0xdc0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xdc8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xdd0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xdd8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x5d,
-/* 0xde0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xde8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x5e,
-/* 0xdf0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x40, 0x5f,
-/* 0xdf8: */  0x00, 0x40, 0x40, 0x5f, 0x5c, 0x5f, 0x5f, 0x5f,
-/* 0xe00: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe08: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe10: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe18: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe20: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe28: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe30: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe38: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe40: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe48: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe50: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe58: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe60: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe68: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe70: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe78: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x67,
-/* 0xe80: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe88: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe90: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe98: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xea0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xea8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xeb0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xeb8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x40, 0x6b,
-/* 0xec0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xec8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xed0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xed8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x40, 0x6d,
-/* 0xee0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x40,
-/* 0xee8: */  0x00, 0x00, 0x00, 0x40, 0x00, 0x40, 0x40, 0x6e,
-/* 0xef0: */  0x00, 0x00, 0x00, 0x40, 0x00, 0x60, 0x60, 0x6f,
-/* 0xef8: */  0x00, 0x60, 0x60, 0x6f, 0x60, 0x6f, 0x6f, 0x6f,
-/* 0xf00: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf08: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf10: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf18: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x40,
-/* 0xf20: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf28: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x40,
-/* 0xf30: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x40,
-/* 0xf38: */  0x00, 0x00, 0x00, 0x40, 0x00, 0x40, 0x60, 0x73,
-/* 0xf40: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf48: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x40,
-/* 0xf50: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x40,
-/* 0xf58: */  0x00, 0x00, 0x00, 0x40, 0x00, 0x60, 0x60, 0x75,
-/* 0xf60: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x60,
-/* 0xf68: */  0x00, 0x00, 0x00, 0x60, 0x00, 0x60, 0x60, 0x76,
-/* 0xf70: */  0x00, 0x00, 0x00, 0x60, 0x00, 0x60, 0x60, 0x77,
-/* 0xf78: */  0x00, 0x70, 0x70, 0x77, 0x70, 0x77, 0x77, 0x77,
-/* 0xf80: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf88: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x60,
-/* 0xf90: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x60,
-/* 0xf98: */  0x00, 0x00, 0x00, 0x60, 0x00, 0x60, 0x60, 0x79,
-/* 0xfa0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x60,
-/* 0xfa8: */  0x00, 0x00, 0x00, 0x60, 0x00, 0x70, 0x70, 0x7a,
-/* 0xfb0: */  0x00, 0x00, 0x00, 0x70, 0x00, 0x70, 0x70, 0x7b,
-/* 0xfb8: */  0x40, 0x70, 0x70, 0x7b, 0x78, 0x7b, 0x7b, 0x7b,
-/* 0xfc0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x70,
-/* 0xfc8: */  0x00, 0x00, 0x00, 0x70, 0x00, 0x70, 0x70, 0x7c,
-/* 0xfd0: */  0x00, 0x00, 0x00, 0x70, 0x40, 0x70, 0x70, 0x7d,
-/* 0xfd8: */  0x40, 0x70, 0x78, 0x7d, 0x78, 0x7d, 0x7d, 0x7d,
-/* 0xfe0: */  0x00, 0x40, 0x40, 0x78, 0x60, 0x78, 0x78, 0x7e,
-/* 0xfe8: */  0x60, 0x78, 0x78, 0x7e, 0x7c, 0x7e, 0x7e, 0x7e,
-/* 0xff0: */  0x70, 0x7c, 0x7c, 0x7f, 0x7c, 0x7f, 0x7f, 0x7f,
-/* 0xff8: */  0x7e, 0x7f, 0x7f, 0x7f, 0x7f, 0x7f, 0x7f, 0x7f,
-};
-
-const reg8 WaveformGenerator::wave6581_PST[] = {
-/* 0x000: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x008: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x010: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x018: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x020: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x028: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x030: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x038: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x040: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x048: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x050: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x058: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x060: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x068: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x070: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x078: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x080: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x088: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x090: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x098: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0e8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0f0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x0f8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x100: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x108: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x110: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x118: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x120: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x128: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x130: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x138: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x140: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x148: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x150: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x158: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x160: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x168: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x170: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x178: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x180: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x188: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x190: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x198: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1e8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1f0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x1f8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x200: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x208: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x210: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x218: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x220: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x228: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x230: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x238: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x240: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x248: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x250: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x258: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x260: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x268: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x270: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x278: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x280: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x288: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x290: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x298: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2e8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2f0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x2f8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x300: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x308: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x310: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x318: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x320: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x328: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x330: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x338: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x340: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x348: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x350: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x358: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x360: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x368: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x370: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x378: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x380: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x388: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x390: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x398: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x3a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x3a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x3b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x3b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x3c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x3c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x3d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x3d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x3e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x3e8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x3f0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x3f8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x3f,
-/* 0x400: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x408: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x410: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x418: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x420: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x428: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x430: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x438: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x440: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x448: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x450: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x458: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x460: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x468: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x470: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x478: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x480: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x488: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x490: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x498: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4e8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4f0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x4f8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x500: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x508: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x510: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x518: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x520: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x528: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x530: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x538: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x540: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x548: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x550: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x558: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x560: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x568: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x570: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x578: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x580: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x588: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x590: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x598: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x5a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x5a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x5b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x5b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x5c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x5c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x5d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x5d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x5e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x5e8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x5f0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x5f8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x600: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x608: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x610: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x618: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x620: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x628: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x630: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x638: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x640: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x648: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x650: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x658: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x660: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x668: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x670: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x678: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x680: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x688: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x690: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x698: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x6a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x6a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x6b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x6b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x6c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x6c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x6d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x6d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x6e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x6e8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x6f0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x6f8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x700: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x708: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x710: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x718: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x720: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x728: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x730: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x738: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x740: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x748: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x750: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x758: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x760: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x768: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x770: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x778: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x780: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x788: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x790: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x798: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x7a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x7a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x7b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x7b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x7c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x7c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x7d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x7d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x7e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x7e8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x20,
-/* 0x7f0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x30,
-/* 0x7f8: */  0x00, 0x00, 0x00, 0x78, 0x78, 0x7e, 0x7f, 0x7f,
-/* 0x800: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x808: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x810: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x818: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x820: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x828: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x830: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x838: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x840: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x848: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x850: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x858: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x860: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x868: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x870: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x878: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x880: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x888: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x890: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x898: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8e8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8f0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x8f8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x900: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x908: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x910: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x918: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x920: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x928: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x930: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x938: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x940: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x948: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x950: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x958: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x960: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x968: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x970: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x978: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x980: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x988: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x990: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x998: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9a0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9a8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9b0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9b8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9c0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9c8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9d0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9d8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9e0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9e8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9f0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0x9f8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa00: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa08: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa10: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa18: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa20: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa28: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa30: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa38: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa40: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa48: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa50: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa58: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa60: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa68: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa70: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa78: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa80: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa88: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa90: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xa98: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xaa0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xaa8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xab0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xab8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xac0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xac8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xad0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xad8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xae0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xae8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xaf0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xaf8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb00: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb08: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb10: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb18: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb20: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb28: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb30: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb38: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb40: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb48: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb50: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb58: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb60: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb68: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb70: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb78: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb80: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb88: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb90: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xb98: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xba0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xba8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbb0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbb8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbc0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbc8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbd0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbd8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbe0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbe8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbf0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xbf8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x3f,
-/* 0xc00: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc08: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc10: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc18: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc20: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc28: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc30: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc38: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc40: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc48: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc50: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc58: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc60: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc68: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc70: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc78: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc80: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc88: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc90: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xc98: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xca0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xca8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xcb0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xcb8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xcc0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xcc8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xcd0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xcd8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xce0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xce8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xcf0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xcf8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd00: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd08: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd10: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd18: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd20: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd28: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd30: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd38: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd40: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd48: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd50: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd58: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd60: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd68: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd70: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd78: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd80: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd88: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd90: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xd98: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xda0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xda8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xdb0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xdb8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xdc0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xdc8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xdd0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xdd8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xde0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xde8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xdf0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xdf8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe00: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe08: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe10: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe18: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe20: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe28: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe30: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe38: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe40: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe48: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe50: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe58: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe60: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe68: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe70: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe78: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe80: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe88: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe90: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xe98: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xea0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xea8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xeb0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xeb8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xec0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xec8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xed0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xed8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xee0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xee8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xef0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xef8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf00: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf08: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf10: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf18: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf20: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf28: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf30: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf38: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf40: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf48: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf50: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf58: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf60: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf68: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf70: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf78: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf80: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf88: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf90: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xf98: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xfa0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xfa8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xfb0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xfb8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xfc0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xfc8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xfd0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xfd8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xfe0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-/* 0xfe8: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x20,
-/* 0xff0: */  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x30,
-/* 0xff8: */  0x00, 0x00, 0x00, 0x78, 0x78, 0x7e, 0x7f, 0x7f,
-};
-
-}
-#endif
diff --git a/sound/softsynth/ym2612.cpp b/sound/softsynth/ym2612.cpp
deleted file mode 100644
index 08331c6244..0000000000
--- a/sound/softsynth/ym2612.cpp
+++ /dev/null
@@ -1,789 +0,0 @@
-/* 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 <math.h>
-
-#include "sound/softsynth/ym2612.h"
-#include "common/util.h"
-#include "sound/musicplugin.h"
-#include "common/translation.h"
-
-////////////////////////////////////////
-//
-// Miscellaneous
-//
-////////////////////////////////////////
-
-static int *sintbl = 0;
-static int *powtbl = 0;
-static int *frequencyTable = 0;
-static int *keycodeTable = 0;
-static int *keyscaleTable = 0;
-static int *attackOut = 0;
-
-
-////////////////////////////////////////
-//
-// Operator2612 implementation
-//
-////////////////////////////////////////
-
-Operator2612::Operator2612 (Voice2612 *owner) :
-	_owner (owner),
-	_state (_s_ready),
-	_currentLevel ((int32)0x7f << 15),
-	_phase (0),
-	_lastOutput (0),
-	_feedbackLevel (0),
-	_detune (0),
-	_multiple (1),
-	_keyScale (0),
-	_specifiedTotalLevel (127),
-	_specifiedAttackRate (0),
-	_specifiedDecayRate (0),
-	_specifiedSustainRate (0),
-	_specifiedReleaseRate (15) {
-		velocity(0);
-}
-
-Operator2612::~Operator2612()
-{ }
-
-void Operator2612::velocity(int velo) {
-	_velocity = velo;
-	_totalLevel = ((int32)_specifiedTotalLevel << 15) +
-					((int32)(127-_velocity) << 13);
-	_sustainLevel = ((int32)_specifiedSustainLevel << 17);
-}
-
-void Operator2612::feedbackLevel(int level) {
-	_feedbackLevel = level;
-}
-
-void Operator2612::setInstrument(byte const *instrument) {
-	_detune = (instrument[8] >> 4) & 7;
-	_multiple = instrument[8] & 15;
-	_specifiedTotalLevel = instrument[12] & 127;
-	_keyScale = (instrument[16] >> 6) & 3;
-	_specifiedAttackRate = instrument[16] & 31;
-	_specifiedDecayRate = instrument[20] & 31;
-	_specifiedSustainRate = instrument[24] & 31;
-	_specifiedSustainLevel = (instrument[28] >> 4) & 15;
-	_specifiedReleaseRate = instrument[28] & 15;
-	_state = _s_ready;
-	velocity(_velocity);
-}
-
-void Operator2612::keyOn() {
-	_state = _s_attacking;
-	_tickCount = 0;
-	_phase = 0;
-	_currentLevel = ((int32)0x7f << 15);
-}
-
-void Operator2612::keyOff() {
-	if (_state != _s_ready)
-		_state = _s_releasing;
-}
-
-void Operator2612::frequency(int freq) {
-	double value; // Use for intermediate computations to avoid int64 arithmetic
-	int r;
-
-	_frequency = freq / _owner->_rate;
-
-	r = _specifiedAttackRate;
-	if (r != 0) {
-		r = r * 2 + (keyscaleTable[freq/262205] >> (3-_keyScale));
-		if (r >= 64)
-		r = 63;
-	}
-
-	r = 63 - r;
-	if (_specifiedTotalLevel >= 128)
-		value = 0;
-	else {
-		value = powtbl[(r&3) << 7];
-		value *= 1 << (r >> 2);
-		value *= 41;
-		value /= 1 << (15 + 5);
-		value *= 127 - _specifiedTotalLevel;
-		value /= 127;
-	}
-	_attackTime = (int32) value; // 1 ?? == (1 << 12)
-	if (_attackTime > 0)
-		_attackTime = (1 << (12+10)) / (_owner->_rate * _attackTime);
-
-	r = _specifiedDecayRate;
-	if (r != 0) {
-		r = r * 2 + (keyscaleTable[freq/262205] >> (3-_keyScale));
-		if (r >= 64)
-			r = 63;
-	}
-	value = (double) powtbl[(r&3) << 7] * (0x10 << (r>>2)) / 31;
-	_decayRate = (int32) value / _owner->_rate;
-
-	r = _specifiedSustainRate;
-	if (r != 0) {
-		r = r * 2 + (keyscaleTable[freq/262205] >> (3-_keyScale));
-		if (r >= 64)
-			r = 63;
-	}
-	value = (double) powtbl[(r&3) << 7] * (0x10 << (r>>2)) / 31;
-	_sustainRate = (int32) value / _owner->_rate;
-
-	r = _specifiedReleaseRate;
-	if (r != 0) {
-		r = r * 2 + 1;		// (Translated) I cannot know whether the timing is a good choice or not
-		r = r * 2 + (keyscaleTable[freq/262205] >> (3-_keyScale));
-		// KS
-		if (r >= 64)
-			r = 63;
-	}
-	value = (double) powtbl[(r&3) << 7] * (0x10 << (r>>2)) / 31;
-	_releaseRate = (int32) value / _owner->_rate;
-}
-
-void Operator2612::nextTick(const int *phasebuf, int *outbuf, int buflen) {
-	if (_state == _s_ready)
-		return;
-	if (_state == _s_attacking && _attackTime <= 0) {
-		_currentLevel = 0;
-		_state = _s_decaying;
-	}
-
-	int32 levelIncrement = 0;
-	int32 target = 0;
-	State next_state = _s_ready;
-	const int32 zero_level = ((int32)0x7f << 15);
-	const int phaseIncrement = (_multiple > 0) ? (_frequency * _multiple) : (_frequency / 2);
-
-	int32 output = _lastOutput;
-	int32 level = _currentLevel + _totalLevel;
-
-	while (buflen) {
-		switch (_state) {
-		case _s_ready:
-			return;
-		case _s_attacking:
-			next_state = _s_attacking;
-			break;
-		case _s_decaying:
-			levelIncrement = _decayRate;
-			target = _sustainLevel + _totalLevel;
-			next_state = _s_sustaining;
-			break;
-		case _s_sustaining:
-			levelIncrement = _sustainRate;
-			target = zero_level + _totalLevel;
-			next_state = _s_ready;
-			break;
-		case _s_releasing:
-			levelIncrement = _releaseRate;
-			target = zero_level + _totalLevel;
-			next_state = _s_ready;
-			break;
-		}
-
-		bool switching = false;
-		do {
-			if (next_state == _s_attacking) {
-				// Attack phase
-				++_tickCount;
-				int i = (int) (_tickCount * _attackTime);
-				if (i >= 1024) {
-					level = _totalLevel;
-					_state = _s_decaying;
-					switching = true;
-				} else {
-					level = (attackOut[i] << (31 - 8 - 16)) + _totalLevel;
-				}
-			} else {
-				// Decay, Sustain and Release phases
-				level += levelIncrement;
-				if (level >= target) {
-					level = target;
-					_state = next_state;
-					switching = true;
-				}
-			}
-
-			if (level < zero_level) {
-				int phaseShift = *phasebuf >> 2;
-				if (_feedbackLevel)
-					phaseShift += (output << (_feedbackLevel - 1)) / 1024;
-				output = sintbl[((_phase >> 7) + phaseShift) & 0x7ff];
-				output >>= (level >> 18);
-				// Here is the original code, which requires 64-bit ints
-//				output *= powtbl[511 - ((level>>25)&511)];
-//				output >>= 16;
-//				output >>= 1;
-				// And here's our 32-bit trick for doing it. (Props to Fingolfin!)
-				// Result varies from original code by max of 1.
-//				int powVal = powtbl[511 - ((level>>9)&511)];
-//				int outputHI = output / 256;
-//				int powHI = powVal / 256;
-//				output = (outputHI * powHI) / 2 + (outputHI * (powVal % 256) + powHI * (output % 256)) / 512;
-				// And here's the even faster code.
-				// Result varies from original code by max of 8.
-				output = ((output >> 4) * (powtbl[511-((level>>9)&511)] >> 3)) / 1024;
-
-				_phase += phaseIncrement;
-				_phase &= 0x3ffff;
-			} else
-				output = 0;
-
-			*outbuf += output;
-			 --buflen;
-			 ++phasebuf;
-			 ++outbuf;
-		} while (buflen && !switching);
-	}
-	_lastOutput = output;
-	_currentLevel = level - _totalLevel;
-}
-
-////////////////////////////////////////
-//
-// Voice2612 implementation
-//
-////////////////////////////////////////
-
-Voice2612::Voice2612() {
-	next = 0;
-	_control7 = 127;
-	_note = 40;
-	_frequency = 440;
-	_frequencyOffs = 0x2000;
-	_algorithm = 7;
-
-	_buffer = 0;
-	_buflen = 0;
-
-	int i;
-	for (i = 0; i < ARRAYSIZE(_opr); ++i)
-		_opr[i] = new Operator2612 (this);
-	velocity(0);
-}
-
-Voice2612::~Voice2612() {
-	int i;
-	for (i = 0; i < ARRAYSIZE(_opr); ++i)
-		delete _opr[i];
-	free(_buffer);
-}
-
-void Voice2612::velocity(int velo) {
-	_velocity = velo;
-#if 0
-	int v = (velo * _control7) >> 7;
-#else
-	int v = velo + (_control7 - 127) * 4;
-#endif
-	bool iscarrier[8][4] = {
-		{ false, false, false,  true, }, //0
-		{ false, false, false,  true, }, //1
-		{ false, false, false,  true, }, //2
-		{ false, false, false,  true, }, //3
-		{ false,  true, false,  true, }, //4
-		{ false,  true,  true,  true, }, //5
-		{ false,  true,  true,  true, }, //6
-		{  true,  true,  true,  true, }, //7
-	};
-	int opr;
-	for (opr = 0; opr < 4; opr++)
-		if (iscarrier[_algorithm][opr])
-			_opr[opr]->velocity(v);
-		else
-			_opr[opr]->velocity(127);
-}
-
-void Voice2612::setControlParameter(int control, int value) {
-	switch (control) {
-	case 7:
-		_control7 = value;
-		velocity(_velocity);
-		break;
-	case 123:
-		// All notes off
-		noteOff(_note);
-	};
-}
-
-void Voice2612::setInstrument(byte const *instrument) {
-	if (instrument == NULL)
-		return;
-
-	_algorithm = instrument[32] & 7;
-	_opr[0]->feedbackLevel((instrument[32] >> 3) & 7);
-	_opr[1]->feedbackLevel(0);
-	_opr[2]->feedbackLevel(0);
-	_opr[3]->feedbackLevel(0);
-	_opr[0]->setInstrument(instrument + 0);
-	_opr[1]->setInstrument(instrument + 2);
-	_opr[2]->setInstrument(instrument + 1);
-	_opr[3]->setInstrument(instrument + 3);
-}
-
-void Voice2612::nextTick(int *outbuf, int buflen) {
-	if (_velocity == 0)
-		return;
-
-	if (_buflen < buflen) {
-		free(_buffer);
-		_buflen = buflen;
-		_buffer = (int *) malloc(sizeof(int) * buflen * 2);
-	}
-
-	int *buf1 = _buffer;
-	int *buf2 = _buffer + buflen;
-	memset(_buffer, 0, sizeof(int) * buflen * 2);
-
-	switch (_algorithm) {
-	case 0:
-		_opr[0]->nextTick(buf1, buf2, buflen);
-		_opr[1]->nextTick(buf2, buf1, buflen);
-		memset (buf2, 0, sizeof (int) * buflen);
-		_opr[2]->nextTick(buf1, buf2, buflen);
-		_opr[3]->nextTick(buf2, outbuf, buflen);
-		break;
-	case 1:
-		_opr[0]->nextTick(buf1, buf2, buflen);
-		_opr[1]->nextTick(buf1, buf2, buflen);
-		_opr[2]->nextTick(buf2, buf1, buflen);
-		_opr[3]->nextTick(buf1, outbuf, buflen);
-		break;
-	case 2:
-		_opr[1]->nextTick(buf1, buf2, buflen);
-		_opr[2]->nextTick(buf2, buf1, buflen);
-		memset(buf2, 0, sizeof(int) * buflen);
-		_opr[0]->nextTick(buf2, buf1, buflen);
-		_opr[3]->nextTick(buf1, outbuf, buflen);
-		break;
-	case 3:
-		_opr[0]->nextTick(buf1, buf2, buflen);
-		_opr[1]->nextTick(buf2, buf1, buflen);
-		memset(buf2, 0, sizeof(int) * buflen);
-		_opr[2]->nextTick(buf2, buf1, buflen);
-		_opr[3]->nextTick(buf1, outbuf, buflen);
-		break;
-	case 4:
-		_opr[0]->nextTick(buf1, buf2, buflen);
-		_opr[1]->nextTick(buf2, outbuf, buflen);
-		_opr[2]->nextTick(buf1, buf1, buflen);
-		_opr[3]->nextTick(buf1, outbuf, buflen);
-		break;
-	case 5:
-		_opr[0]->nextTick(buf1, buf2, buflen);
-		_opr[1]->nextTick(buf2, outbuf, buflen);
-		_opr[2]->nextTick(buf2, outbuf, buflen);
-		_opr[3]->nextTick(buf2, outbuf, buflen);
-		break;
-	case 6:
-		_opr[0]->nextTick(buf1, buf2, buflen);
-		_opr[1]->nextTick(buf2, outbuf, buflen);
-		_opr[2]->nextTick(buf1, outbuf, buflen);
-		_opr[3]->nextTick(buf1, outbuf, buflen);
-		break;
-	case 7:
-		_opr[0]->nextTick(buf1, outbuf, buflen);
-		_opr[1]->nextTick(buf1, outbuf, buflen);
-		_opr[2]->nextTick(buf1, outbuf, buflen);
-		_opr[3]->nextTick(buf1, outbuf, buflen);
-		break;
-	};
-}
-
-void Voice2612::noteOn(int n, int onVelo) {
-	_note = n;
-	velocity(onVelo);
-	recalculateFrequency();
-	int i;
-	for (i = 0; i < ARRAYSIZE(_opr); i++)
-		_opr[i]->keyOn();
-}
-
-bool Voice2612::noteOff(int note) {
-	if (_note != note)
-		return false;
-	int i;
-	for (i = 0; i < ARRAYSIZE(_opr); i++)
-		_opr[i]->keyOff();
-	return true;
-}
-
-void Voice2612::pitchBend(int value) {
-	_frequencyOffs = value;
-	recalculateFrequency();
-}
-
-void Voice2612::recalculateFrequency() {
-	//
-	//
-	//
-	int32 basefreq = frequencyTable[_note];
-	int cfreq = frequencyTable[_note - (_note % 12)];
-	int oct = _note / 12;
-	int fnum = (int) (((double)basefreq * (1 << 13)) / cfreq);
-	fnum += _frequencyOffs - 0x2000;
-	if (fnum < 0x2000) {
-		fnum += 0x2000;
-		oct--;
-	}
-	if (fnum >= 0x4000) {
-		fnum -= 0x2000;
-		oct++;
-	}
-
-	//
-	_frequency = (int) ((frequencyTable[oct*12] * (double)fnum) / 8);
-
-	int i;
-	for (i = 0; i < ARRAYSIZE(_opr); i++)
-		_opr[i]->frequency(_frequency);
-}
-
-////////////////////////////////////////
-//
-// MidiChannel_YM2612
-//
-////////////////////////////////////////
-
-MidiChannel_YM2612::MidiChannel_YM2612() {
-	_voices = 0;
-	_next_voice = 0;
-}
-
-MidiChannel_YM2612::~MidiChannel_YM2612() {
-	removeAllVoices();
-}
-
-void MidiChannel_YM2612::removeAllVoices() {
-	if (!_voices)
-		return;
-	Voice2612 *last, *voice = _voices;
-	for (; voice; voice = last) {
-		last = voice->next;
-		delete voice;
-	}
-	_voices = _next_voice = 0;
-}
-
-void MidiChannel_YM2612::noteOn(byte note, byte onVelo) {
-	if (!_voices)
-		return;
-	_next_voice = _next_voice ? _next_voice : _voices;
-	_next_voice->noteOn(note, onVelo);
-	_next_voice = _next_voice->next;
-}
-
-void MidiChannel_YM2612::noteOff(byte note) {
-	if (!_voices)
-		return;
-	if (_next_voice == _voices)
-		_next_voice = 0;
-	Voice2612 *voice = _next_voice;
-	do {
-		if (!voice)
-			voice = _voices;
-		if (voice->noteOff(note)) {
-			_next_voice = voice;
-			break;
-		}
-		voice = voice->next;
-	} while (voice != _next_voice);
-}
-
-void MidiChannel_YM2612::controlChange(byte control, byte value) {
-	//
-	if (control == 121) {
-		// Reset controller
-		removeAllVoices();
-	} else {
-		Voice2612 *voice = _voices;
-		for (; voice; voice = voice->next)
-			voice->setControlParameter(control, value);
-	}
-}
-
-void MidiChannel_YM2612::sysEx_customInstrument(uint32 type, const byte *fmInst) {
-	if (type != 'EUP ')
-		return;
-	Voice2612 *voice = new Voice2612;
-	voice->next = _voices;
-	_voices = voice;
-	voice->_rate = _rate;
-	voice->setInstrument(fmInst);
-}
-
-void MidiChannel_YM2612::pitchBend(int16 value) {
-  //
-	Voice2612 *voice = _voices;
-	for (; voice; voice = voice->next)
-		voice->pitchBend(value);
-}
-
-void MidiChannel_YM2612::nextTick(int *outbuf, int buflen) {
-	Voice2612 *voice = _voices;
-	for (; voice; voice = voice->next)
-		voice->nextTick(outbuf, buflen);
-}
-
-void MidiChannel_YM2612::rate(uint16 r) {
-	_rate = r;
-	Voice2612 *voice = _voices;
-	for (; voice; voice = voice->next)
-		voice->_rate = r;
-}
-
-////////////////////////////////////////
-//
-// MidiDriver_YM2612
-//
-////////////////////////////////////////
-
-MidiDriver_YM2612::MidiDriver_YM2612(Audio::Mixer *mixer)
-	: MidiDriver_Emulated(mixer) {
-	_next_voice = 0;
-
-	createLookupTables();
-	_volume = 256;
-	int i;
-	for (i = 0; i < ARRAYSIZE(_channel); i++)
-		_channel[i] = new MidiChannel_YM2612;
-	rate(getRate());
-}
-
-MidiDriver_YM2612::~MidiDriver_YM2612() {
-	int i;
-	for (i = 0; i < ARRAYSIZE(_channel); i++)
-		delete _channel[i];
-	removeLookupTables();
-}
-
-int MidiDriver_YM2612::open() {
-	if (_isOpen)
-		return MERR_ALREADY_OPEN;
-
-	MidiDriver_Emulated::open();
-
-	_mixer->playStream(Audio::Mixer::kPlainSoundType, &_mixerSoundHandle, this, -1, Audio::Mixer::kMaxChannelVolume, 0, DisposeAfterUse::NO, true);
-	return 0;
-}
-
-void MidiDriver_YM2612::close() {
-	if (!_isOpen)
-		return;
-	_isOpen = false;
-
-	_mixer->stopHandle(_mixerSoundHandle);
-}
-
-void MidiDriver_YM2612::send(uint32 b) {
-	send(b & 0xF, b & 0xFFFFFFF0);
-}
-
-void MidiDriver_YM2612::send(byte chan, uint32 b) {
-	//byte param3 = (byte) ((b >> 24) & 0xFF);
-	byte param2 = (byte) ((b >> 16) & 0xFF);
-	byte param1 = (byte) ((b >>  8) & 0xFF);
-	byte cmd    = (byte) (b & 0xF0);
-	if (chan > ARRAYSIZE(_channel))
-		return;
-
-	switch (cmd) {
-	case 0x80:// Note Off
-		_channel[chan]->noteOff(param1);
-		break;
-	case 0x90: // Note On
-		_channel[chan]->noteOn(param1, param2);
-		break;
-	case 0xA0: // Aftertouch
-		break; // Not supported.
-	case 0xB0: // Control Change
-		_channel[chan]->controlChange(param1, param2);
-		break;
-	case 0xC0: // Program Change
-		_channel[chan]->programChange(param1);
-		break;
-	case 0xD0: // Channel Pressure
-		break; // Not supported.
-	case 0xE0: // Pitch Bend
-		_channel[chan]->pitchBend((param1 | (param2 << 7)) - 0x2000);
-		break;
-	case 0xF0: // SysEx
-		// We should never get here! SysEx information has to be
-		// sent via high-level semantic methods.
-		warning("MidiDriver_YM2612: Receiving SysEx command on a send() call");
-		break;
-
-	default:
-		warning("MidiDriver_YM2612: Unknown send() command 0x%02X", cmd);
-	}
-}
-
-void MidiDriver_YM2612::sysEx(const byte *msg, uint16 length) {
-	if (msg[0] != 0x7C || msg[1] >= ARRAYSIZE(_channel))
-		return;
-	_channel[msg[1]]->sysEx_customInstrument('EUP ', &msg[2]);
-}
-
-void MidiDriver_YM2612::generateSamples(int16 *data, int len) {
-	memset(data, 0, 2 * sizeof(int16) * len);
-	nextTick(data, len);
-}
-
-void MidiDriver_YM2612::nextTick(int16 *buf1, int buflen) {
-	int *buf0 = (int *)buf1;
-
-	int i;
-	for (i = 0; i < ARRAYSIZE(_channel); i++)
-		_channel[i]->nextTick(buf0, buflen);
-
-	for (i = 0; i < buflen; ++i)
-		buf1[i*2+1] = buf1[i*2] = ((buf0[i] * volume()) >> 10) & 0xffff;
-}
-
-void MidiDriver_YM2612::rate(uint16 r)
-{
-	int i;
-	for (i = 0; i < ARRAYSIZE(_channel); i++)
-		_channel[i]->rate(r);
-}
-
-void MidiDriver_YM2612::createLookupTables() {
-	{
-		int i;
-		sintbl = new int [2048];
-		for (i = 0; i < 2048; i++)
-			sintbl[i] = (int)(0xffff * sin(i/2048.0*2.0*PI));
-	}
-
-	{
-		int i;
-		powtbl = new int [1025];
-		for (i = 0; i <= 1024; i++)
-			powtbl[i] = (int)(0x10000 * pow(2.0, (i-512)/512.0));
-	}
-
-	{
-		int i;
-		int block;
-
-		static int fnum[] = {
-			0x026a, 0x028f, 0x02b6, 0x02df,
-			0x030b, 0x0339, 0x036a, 0x039e,
-			0x03d5, 0x0410, 0x044e, 0x048f,
-		};
-
-		// (int)(880.0 * 256.0 * pow(2.0, (note-0x51)/12.0))
-		//
-		frequencyTable = new int [120];
-		for (block = -1; block < 9; block++) {
-			for (i = 0; i < 12; i++) {
-				double freq = fnum[i] * (166400.0 / 3) * pow(2.0, block-21);
-				frequencyTable[(block+1)*12+i] = (int)(256.0 * freq);
-			}
-		}
-
-		keycodeTable = new int [120];
-		// detune
-		for (block = -1; block < 9; block++) {
-			for (i = 0; i < 12; i++) {
-				// see p.204
-				int  f8 = (fnum[i] >>  7) & 1;
-				int  f9 = (fnum[i] >>  8) & 1;
-				int f10 = (fnum[i] >>  9) & 1;
-				int f11 = (fnum[i] >> 10) & 1;
-				int  n4 = f11;
-				int  n3 = (f11&(f10|f9|f8)) | (~f11&f10&f9&f8);
-				int note = n4*2 + n3;
-				// see p.207
-				keycodeTable[(block+1)*12+i] = block*4 + note;
-			}
-		}
-	}
-
-	{
-		int freq;
-		keyscaleTable = new int [8192];
-		keyscaleTable[0] = 0;
-		for (freq = 1; freq < 8192; freq++) {
-			keyscaleTable[freq] = (int)(log((double)freq) / 9.03 * 32.0) - 1;
-			// 8368[Hz] (o9c)
-		}
-	}
-
-	{
-		int i;
-		attackOut = new int [1024];
-		for (i = 0; i < 1024; i++)
-			attackOut[i] = (int)(((0x7fff+0x03a5)*30.0) / (30.0+i)) - 0x03a5;
-	}
-}
-
-void MidiDriver_YM2612::removeLookupTables() {
-	delete[] sintbl;
-	delete[] powtbl;
-	delete[] frequencyTable;
-	delete[] keycodeTable;
-	delete[] keyscaleTable;
-	delete[] attackOut;
-	sintbl = powtbl = frequencyTable = keycodeTable = keyscaleTable = attackOut = 0;
-}
-
-
-// Plugin interface
-
-class TownsEmuMusicPlugin : public MusicPluginObject {
-public:
-	const char *getName() const {
-		return _s("FM Towns Emulator");
-	}
-
-	const char *getId() const {
-		return "towns";
-	}
-
-	MusicDevices getDevices() const;
-	Common::Error createInstance(MidiDriver **mididriver, MidiDriver::DeviceHandle = 0) const;
-};
-
-MusicDevices TownsEmuMusicPlugin::getDevices() const {
-	MusicDevices devices;
-	devices.push_back(MusicDevice(this, "", MT_TOWNS));
-	return devices;
-}
-
-Common::Error TownsEmuMusicPlugin::createInstance(MidiDriver **mididriver, MidiDriver::DeviceHandle) const {
-	*mididriver = new MidiDriver_YM2612(g_system->getMixer());
-
-	return Common::kNoError;
-}
-
-//#if PLUGIN_ENABLED_DYNAMIC(TOWNS)
-	//REGISTER_PLUGIN_DYNAMIC(TOWNS, PLUGIN_TYPE_MUSIC, TownsEmuMusicPlugin);
-//#else
-	REGISTER_PLUGIN_STATIC(TOWNS, PLUGIN_TYPE_MUSIC, TownsEmuMusicPlugin);
-//#endif
diff --git a/sound/softsynth/ym2612.h b/sound/softsynth/ym2612.h
deleted file mode 100644
index 5fb2f32ef0..0000000000
--- a/sound/softsynth/ym2612.h
+++ /dev/null
@@ -1,179 +0,0 @@
-/* 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_SOFTSYNTH_Y2612_H
-#define SOUND_SOFTSYNTH_Y2612_H
-
-#include "common/scummsys.h"
-
-#include "sound/softsynth/emumidi.h"
-
-////////////////////////////////////////
-//
-// Class declarations
-//
-////////////////////////////////////////
-
-class Voice2612;
-class Operator2612 {
-protected:
-	Voice2612 *_owner;
-	enum State { _s_ready, _s_attacking, _s_decaying, _s_sustaining, _s_releasing };
-	State _state;
-	int32 _currentLevel;
-	int _frequency;
-	uint32 _phase;
-	int _lastOutput;
-	int _feedbackLevel;
-	int _detune;
-	int _multiple;
-	int32 _totalLevel;
-	int _keyScale;
-	int _velocity;
-	int _specifiedTotalLevel;
-	int _specifiedAttackRate;
-	int _specifiedDecayRate;
-	int _specifiedSustainLevel;
-	int _specifiedSustainRate;
-	int _specifiedReleaseRate;
-	int _tickCount;
-	int _attackTime;
-	int32 _decayRate;
-	int32 _sustainLevel;
-	int32 _sustainRate;
-	int32 _releaseRate;
-
-public:
-	Operator2612 (Voice2612 *owner);
-	~Operator2612();
-	void feedbackLevel(int level);
-	void setInstrument(byte const *instrument);
-	void velocity(int velo);
-	void keyOn();
-	void keyOff();
-	void frequency(int freq);
-	void nextTick(const int *phaseShift, int *outbuf, int buflen);
-	bool inUse() { return (_state != _s_ready); }
-};
-
-class Voice2612 {
-public:
-	Voice2612 *next;
-	uint16 _rate;
-
-protected:
-	Operator2612 *_opr[4];
-	int _velocity;
-	int _control7;
-	int _note;
-	int _frequencyOffs;
-	int _frequency;
-	int _algorithm;
-
-	int *_buffer;
-	int _buflen;
-
-public:
-	Voice2612();
-	~Voice2612();
-	void setControlParameter(int control, int value);
-	void setInstrument(byte const *instrument);
-	void velocity(int velo);
-	void nextTick(int *outbuf, int buflen);
-	void noteOn(int n, int onVelo);
-	bool noteOff(int note);
-	void pitchBend(int value);
-	void recalculateFrequency();
-};
-
-class MidiChannel_YM2612 : public MidiChannel {
-protected:
-	uint16 _rate;
-	Voice2612 *_voices;
-	Voice2612 *_next_voice;
-
-public:
-	void removeAllVoices();
-	void nextTick(int *outbuf, int buflen);
-	void rate(uint16 r);
-
-public:
-	MidiChannel_YM2612();
-	virtual ~MidiChannel_YM2612();
-
-	// MidiChannel interface
-	MidiDriver *device() { return 0; }
-	byte getNumber() { return 0; }
-	void release() { }
-	void send(uint32 b) { }
-	void noteOff(byte note);
-	void noteOn(byte note, byte onVelo);
-	void programChange(byte program) { }
-	void pitchBend(int16 value);
-	void controlChange(byte control, byte value);
-	void pitchBendFactor(byte value) { }
-	void sysEx_customInstrument(uint32 type, const byte *instr);
-};
-
-class MidiDriver_YM2612 : public MidiDriver_Emulated {
-protected:
-	MidiChannel_YM2612 *_channel[16];
-
-	int _next_voice;
-	int _volume;
-
-protected:
-	void nextTick(int16 *buf1, int buflen);
-	int volume(int val = -1) { if (val >= 0) _volume = val; return _volume; }
-	void rate(uint16 r);
-
-	void generateSamples(int16 *buf, int len);
-
-public:
-	MidiDriver_YM2612(Audio::Mixer *mixer);
-	virtual ~MidiDriver_YM2612();
-
-	static void createLookupTables();
-	static void removeLookupTables();
-
-	int open();
-	void close();
-	void send(uint32 b);
-	void send(byte channel, uint32 b); // Supports higher than channel 15
-	uint32 property(int prop, uint32 param) { return 0; }
-
-	void setPitchBendRange(byte channel, uint range) { }
-	void sysEx(const byte *msg, uint16 length);
-
-	MidiChannel *allocateChannel() { return 0; }
-	MidiChannel *getPercussionChannel() { return 0; }
-
-
-	// AudioStream API
-	bool isStereo() const { return true; }
-	int getRate() const { return _mixer->getOutputRate(); }
-};
-
-#endif
-
diff --git a/sound/timestamp.cpp b/sound/timestamp.cpp
deleted file mode 100644
index 1e49e1b476..0000000000
--- a/sound/timestamp.cpp
+++ /dev/null
@@ -1,212 +0,0 @@
-/* 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 "sound/timestamp.h"
-#include "common/algorithm.h"
-
-namespace Audio {
-
-Timestamp::Timestamp(uint ms, uint fr) {
-	assert(fr > 0);
-
-	_secs = ms / 1000;
-	_framerateFactor = 1000 / Common::gcd<uint>(1000, fr);
-	_framerate = fr * _framerateFactor;
-
-	// Note that _framerate is always divisible by 1000.
-	_numFrames = (ms % 1000) * (_framerate / 1000);
-}
-
-Timestamp::Timestamp(uint s, uint frames, uint fr) {
-	assert(fr > 0);
-
-	_secs = s;
-	_framerateFactor = 1000 / Common::gcd<uint>(1000, fr);
-	_framerate = fr * _framerateFactor;
-	_numFrames = frames * _framerateFactor;
-
-	normalize();
-}
-
-Timestamp Timestamp::convertToFramerate(uint newFramerate) const {
-	Timestamp ts(*this);
-
-	if (ts.framerate() != newFramerate) {
-		ts._framerateFactor = 1000 / Common::gcd<uint>(1000, newFramerate);
-		ts._framerate = newFramerate * ts._framerateFactor;
-
-		const uint g = Common::gcd(_framerate, ts._framerate);
-		const uint p = _framerate / g;
-		const uint q = ts._framerate / g;
-
-		// Convert the frame offset to the new framerate.
-		// We round to the nearest (as opposed to always
-		// rounding down), to minimize rounding errors during
-		// round trip conversions.
-		ts._numFrames = (ts._numFrames * q + p/2) / p;
-
-		ts.normalize();
-	}
-
-	return ts;
-}
-
-void Timestamp::normalize() {
-	// Convert negative _numFrames values to positive ones by adjusting _secs
-	if (_numFrames < 0) {
-		int secsub = 1 + (-_numFrames / _framerate);
-
-		_numFrames += _framerate * secsub;
-		_secs -= secsub;
-	}
-
-	// Wrap around if necessary
-	_secs += (_numFrames / _framerate);
-	_numFrames %= _framerate;
-}
-
-bool Timestamp::operator==(const Timestamp &ts) const {
-	return cmp(ts) == 0;
-}
-
-bool Timestamp::operator!=(const Timestamp &ts) const {
-	return cmp(ts) != 0;
-}
-
-bool Timestamp::operator<(const Timestamp &ts) const {
-	return cmp(ts) < 0;
-}
-
-bool Timestamp::operator<=(const Timestamp &ts) const {
-	return cmp(ts) <= 0;
-}
-
-bool Timestamp::operator>(const Timestamp &ts) const {
-	return cmp(ts) > 0;
-}
-
-bool Timestamp::operator>=(const Timestamp &ts) const {
-	return cmp(ts) >= 0;
-}
-
-int Timestamp::cmp(const Timestamp &ts) const {
-	int delta = _secs - ts._secs;
-	if (!delta) {
-		const uint g = Common::gcd(_framerate, ts._framerate);
-		const uint p = _framerate / g;
-		const uint q = ts._framerate / g;
-
-		delta = (_numFrames * q - ts._numFrames * p);
-	}
-
-	return delta;
-}
-
-
-Timestamp Timestamp::addFrames(int frames) const {
-	Timestamp ts(*this);
-
-	// The frames are given in the original framerate, so we have to
-	// adjust by _framerateFactor accordingly.
-	ts._numFrames += frames * _framerateFactor;
-	ts.normalize();
-
-	return ts;
-}
-
-Timestamp Timestamp::addMsecs(int ms) const {
-	Timestamp ts(*this);
-	ts._secs += ms / 1000;
-	// Add the remaining frames. Note that _framerate is always divisible by 1000.
-	ts._numFrames += (ms % 1000) * (ts._framerate / 1000);
-
-	ts.normalize();
-
-	return ts;
-}
-
-void Timestamp::addIntern(const Timestamp &ts) {
-	assert(_framerate == ts._framerate);
-	_secs += ts._secs;
-	_numFrames += ts._numFrames;
-
-	normalize();
-}
-
-Timestamp Timestamp::operator-() const {
-	Timestamp result(*this);
-	result._secs = -_secs;
-	result._numFrames = -_numFrames;
-	result.normalize();
-	return result;
-}
-
-Timestamp Timestamp::operator+(const Timestamp &ts) const {
-	Timestamp result(*this);
-	result.addIntern(ts);
-	return result;
-}
-
-Timestamp Timestamp::operator-(const Timestamp &ts) const {
-	Timestamp result(*this);
-	result.addIntern(-ts);
-	return result;
-}
-
-int Timestamp::frameDiff(const Timestamp &ts) const {
-
-	int delta = 0;
-	if (_secs != ts._secs)
-		delta = (_secs - ts._secs) * _framerate;
-
-	delta += _numFrames;
-
-	if (_framerate == ts._framerate) {
-		delta -= ts._numFrames;
-	} else {
-		// We need to multiply by the quotient of the two framerates.
-		// We cancel the GCD in this fraction to reduce the risk of
-		// overflows.
-		const uint g = Common::gcd(_framerate, ts._framerate);
-		const uint p = _framerate / g;
-		const uint q = ts._framerate / g;
-
-		delta -= ((long)ts._numFrames * p + q/2) / (long)q;
-	}
-
-	return delta / (int)_framerateFactor;
-}
-
-int Timestamp::msecsDiff(const Timestamp &ts) const {
-	return msecs() - ts.msecs();
-}
-
-int Timestamp::msecs() const {
-	// Note that _framerate is always divisible by 1000.
-	return _secs * 1000 + _numFrames / (_framerate / 1000);
-}
-
-
-} // End of namespace Audio
diff --git a/sound/timestamp.h b/sound/timestamp.h
deleted file mode 100644
index 4130793fc8..0000000000
--- a/sound/timestamp.h
+++ /dev/null
@@ -1,251 +0,0 @@
-/* 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_TIMESTAMP_H
-#define SOUND_TIMESTAMP_H
-
-#include "common/scummsys.h"
-
-namespace Audio {
-
-/**
- * Timestamps allow specifying points in time and measuring time intervals
- * with a sub-millisecond granularity.
- *
- * When dealing with audio and video decoding, it is often necessary to
- * measure time (intervals) in terms of frames, relative to a fixed
- * frame rate (that is, a fixed number of frames per seconds). For
- * example, in a typical video there are 24 frames per second, and in a
- * typical sound there are 44100 frames (i.e. samples for mono sound
- * and pairs of samples for stereo) per second.
- *
- * At the same time, the system clock provided by ScummVM measures time
- * in milliseconds. For syncing purposes and other reasons, it is often
- * necessary to convert between and compare time measures given on the
- * one hand as a frame count, and on the other hand as a number of
- * milliseconds.
- *
- * If handled carelessly, this can introduce rounding errors that
- * quickly accumulate, resulting in user noticeable disturbance, such as
- * audio and video running out of sync. E.g. a typical approach is to
- * measure all time in milliseconds. But with a frame rate of 24 frames
- * per second, one frame is 41.66666... milliseconds long. On the other
- * hand, if measuring in frames, then similar rounding issue occur when
- * converting from milliseconds to frames.
- *
- * One solution is to use floating point arithmetic to compute with
- * fractional frames resp. (milli)seconds. This has other undesirable
- * side effects; foremost, some platforms ScummVM runs on still have
- * only limited (and slow) floating point support.
- *
- * This class provides an alternate solution: It stores time in terms of
- * frames, but with a twist: Client code can specify arbitrary
- * (integral) framerates; but internally, Timestamp modifies the
- * framerate to be a multiple of 1000. This way, both numbers of frames
- * (relative to the original framerate) as well as milliseconds can be
- * represented as integers. This change is completely hidden from the
- * user, however.
- *
- * A Timestamp can be converted to a frame count or milliseconds at
- * virtually no cost. Likewise, it is posible to compute the difference
- * between two Timestamps in milliseconds or number of frames.
- * Timestamps can be easily compared using regular comparison operators,
- * resulting in nicely readable code; this is even possible for
- * timestamps that are specified using different framerates.
- * Client code can modify Timestamps by adding a number of frames
- * to it, or adding a number of milliseconds. Adding negative amounts is
- * also allowed, and a Timestamp can even represent a "negative time"
- * (mainly useful when using the Timestamp to store a time interval).
- */
-class Timestamp {
-public:
-	/**
-	 * Set up a timestamp with a given time and framerate.
-	 * @param msecs     starting time in milliseconds
-	 * @param framerate number of frames per second (must be > 0)
-	 */
-	Timestamp(uint msecs = 0, uint framerate = 1);
-
-	/**
-	 * Set up a timestamp with a given time, frames and framerate.
-	 * @param secs      starting time in seconds
-	 * @param frames    starting frames
-	 * @param framerate number of frames per second (must be > 0)
-	 */
-	Timestamp(uint secs, uint frames, uint framerate);
-
-	/**
-	 * Return a timestamp which represents as closely as possible
-	 * the point in time describes by this timestamp, but with
-	 * a different framerate.
-	 */
-	Timestamp convertToFramerate(uint newFramerate) const;
-
-	/**
-	 * Check whether to timestamps describe the exact same moment
-	 * in time. This means that two timestamps can compare
-	 * as equal even if they use different framerates.
-	 */
-	bool operator==(const Timestamp &ts) const;
-	bool operator!=(const Timestamp &ts) const;
-	bool operator<(const Timestamp &ts) const;
-	bool operator<=(const Timestamp &ts) const;
-	bool operator>(const Timestamp &ts) const;
-	bool operator>=(const Timestamp &ts) const;
-
-	/**
-	 * Returns a new timestamp, which corresponds to the time encoded
-	 * by this timestamp with the given number of frames added.
-	 * @param frames     number of frames to add
-	 */
-	Timestamp addFrames(int frames) const;
-
-	/**
-	 * Returns a new timestamp, which corresponds to the time encoded
-	 * by this timestamp with the given number of milliseconds added.
-	 * @param msecs     number of milliseconds to add
-	 */
-	Timestamp addMsecs(int msecs) const;
-
-
-	// unary minus
-	Timestamp operator-() const;
-
-	/**
-	 * Compute the sum of two timestamps. This is only
-	 * allowed if they use the same framerate.
-	 */
-	Timestamp operator+(const Timestamp &ts) const;
-
-	/**
-	 * Compute the difference between two timestamps. This is only
-	 * allowed if they use the same framerate.
-	 */
-	Timestamp operator-(const Timestamp &ts) const;
-
-	/**
-	 * Computes the number of frames between this timestamp and ts.
-	 * The frames are with respect to the framerate used by this
-	 * Timestamp (which may differ from the framerate used by ts).
-	 */
-	int frameDiff(const Timestamp &ts) const;
-
-	/** Computes the number off milliseconds between this timestamp and ts. */
-	int msecsDiff(const Timestamp &ts) const;
-
-	/**
-	 * Return the time in milliseconds described by this timestamp,
-	 * rounded down.
-	 */
-	int msecs() const;
-
-	/**
-	 * Return the time in seconds described by this timestamp,
-	 * rounded down.
-	 */
-	inline int secs() const {
-		return _secs;
-	}
-
-	/**
-	 * Return the time in frames described by this timestamp.
-	 */
-	inline int totalNumberOfFrames() const {
-		return _numFrames / (int)_framerateFactor + _secs * (int)(_framerate / _framerateFactor);
-	}
-
-	/**
-	 * A timestamp consists of a number of seconds, plus a number
-	 * of frames, the latter describing a fraction of a second.
-	 * This method returns the latter number.
-	 */
-	inline int numberOfFrames() const {
-		return _numFrames / (int)_framerateFactor;
-	}
-
-	/** Return the framerate used by this timestamp. */
-	inline uint framerate() const { return _framerate / _framerateFactor; }
-
-protected:
-	/**
-	 * Compare this timestamp to another one and return
-	 * a value similar to strcmp.
-	 */
-	int cmp(const Timestamp &ts) const;
-
-	/**
-	 * Normalize this timestamp by making _numFrames non-negative
-	 * and reducing it modulo _framerate.
-	 */
-	void normalize();
-
-	/**
-	 * Add another timestamp to this one and normalize the result.
-	 */
-	void addIntern(const Timestamp &ts);
-
-protected:
-	/**
-	 * The seconds part of this timestamp.
-	 * The total time in seconds represented by this timestamp can be
-	 * computed as follows:
-	 *   _secs + (double)_numFrames / _framerate
-	 */
-	int _secs;
-
-	/**
-	 * The number of frames which together with _secs encodes the
-	 * timestamp. The total number of *internal* frames represented
-	 * by this timestamp can be computed as follows:
-	 *   _numFrames + _secs * _framerate
-	 * To obtain the number of frames with respect to the original
-	 * framerate, this value has to be divided by _framerateFactor.
-	 *
-	 * This is always a value greater or equal to zero.
-	 * The only reason this is an int and not an uint is to
-	 * allow intermediate negative values.
-	 */
-	int _numFrames;
-
-	/**
-	 * The internal framerate, i.e. the number of frames per second.
-	 * This is computed as the least common multiple of the framerate
-	 * specified by the client code, and 1000.
-	 * This way, we ensure that we can store both frames and
-	 * milliseconds without any rounding losses.
-	 */
-	uint _framerate;
-
-	/**
-	 * Factor by which the original framerate specified by the client
-	 * code was multipled to obtain the internal _framerate value.
-	 */
-	uint _framerateFactor;
-};
-
-
-} // End of namespace Audio
-
-#endif