Merge branch 'master'

46 files changed, 390 insertions, 487 deletions

diff --git a/audio/audiostream.cpp b/audio/audiostream.cpp
index 547aa77526..1c5c435359 100644
--- a/audio/audiostream.cpp
+++ b/audio/audiostream.cpp
@@ -61,15 +61,13 @@ static const StreamFileFormat STREAM_FILEFORMATS[] = {
 	{ "MPEG Layer 3", ".mp3",  makeMP3Stream },
 #endif
 	{ "MPEG-4 Audio",   ".m4a",  makeQuickTimeStream },
-
-	{ 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) {
+	for (int i = 0; i < ARRAYSIZE(STREAM_FILEFORMATS); ++i) {
 		Common::String filename = basename + STREAM_FILEFORMATS[i].fileExtension;
 		fileHandle->open(filename);
 		if (fileHandle->isOpen()) {
@@ -93,7 +91,7 @@ SeekableAudioStream *SeekableAudioStream::openStreamFile(const Common::String &b
 #pragma mark -
 
 LoopingAudioStream::LoopingAudioStream(RewindableAudioStream *stream, uint loops, DisposeAfterUse::Flag disposeAfterUse)
-    : _parent(stream), _disposeAfterUse(disposeAfterUse), _loops(loops), _completeIterations(0) {
+    : _parent(stream, disposeAfterUse), _loops(loops), _completeIterations(0) {
 	assert(stream);
 
 	if (!stream->rewind()) {
@@ -102,11 +100,6 @@ LoopingAudioStream::LoopingAudioStream(RewindableAudioStream *stream, uint loops
 	}
 }
 
-LoopingAudioStream::~LoopingAudioStream() {
-	if (_disposeAfterUse == DisposeAfterUse::YES)
-		delete _parent;
-}
-
 int LoopingAudioStream::readBuffer(int16 *buffer, const int numSamples) {
 	if ((_loops && _completeIterations == _loops) || !numSamples)
 		return 0;
@@ -169,7 +162,7 @@ SubLoopingAudioStream::SubLoopingAudioStream(SeekableAudioStream *stream,
                                              const Timestamp loopStart,
                                              const Timestamp loopEnd,
                                              DisposeAfterUse::Flag disposeAfterUse)
-    : _parent(stream), _disposeAfterUse(disposeAfterUse), _loops(loops),
+    : _parent(stream, disposeAfterUse), _loops(loops),
       _pos(0, getRate() * (isStereo() ? 2 : 1)),
       _loopStart(convertTimeToStreamPos(loopStart, getRate(), isStereo())),
       _loopEnd(convertTimeToStreamPos(loopEnd, getRate(), isStereo())),
@@ -180,11 +173,6 @@ SubLoopingAudioStream::SubLoopingAudioStream(SeekableAudioStream *stream,
 		_done = true;
 }
 
-SubLoopingAudioStream::~SubLoopingAudioStream() {
-	if (_disposeAfterUse == DisposeAfterUse::YES)
-		delete _parent;
-}
-
 int SubLoopingAudioStream::readBuffer(int16 *buffer, const int numSamples) {
 	if (_done)
 		return 0;
@@ -225,7 +213,7 @@ int SubLoopingAudioStream::readBuffer(int16 *buffer, const int numSamples) {
 #pragma mark -
 
 SubSeekableAudioStream::SubSeekableAudioStream(SeekableAudioStream *parent, const Timestamp start, const Timestamp end, DisposeAfterUse::Flag disposeAfterUse)
-    : _parent(parent), _disposeAfterUse(disposeAfterUse),
+    : _parent(parent, disposeAfterUse),
       _start(convertTimeToStreamPos(start, getRate(), isStereo())),
       _pos(0, getRate() * (isStereo() ? 2 : 1)),
       _length(convertTimeToStreamPos(end, getRate(), isStereo()) - _start) {
@@ -234,11 +222,6 @@ SubSeekableAudioStream::SubSeekableAudioStream(SeekableAudioStream *parent, cons
 	_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);
diff --git a/audio/audiostream.h b/audio/audiostream.h
index 0ffaa241ce..9c28e4d67f 100644
--- a/audio/audiostream.h
+++ b/audio/audiostream.h
@@ -23,6 +23,7 @@
 #ifndef SOUND_AUDIOSTREAM_H
 #define SOUND_AUDIOSTREAM_H
 
+#include "common/ptr.h"
 #include "common/scummsys.h"
 #include "common/str.h"
 #include "common/types.h"
@@ -114,7 +115,6 @@ public:
 	 * @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;
@@ -129,8 +129,7 @@ public:
 	 */
 	uint getCompleteIterations() const { return _completeIterations; }
 private:
-	RewindableAudioStream *_parent;
-	DisposeAfterUse::Flag _disposeAfterUse;
+	Common::DisposablePtr<RewindableAudioStream> _parent;
 
 	uint _loops;
 	uint _completeIterations;
@@ -246,7 +245,6 @@ public:
 	                      const Timestamp loopStart,
 	                      const Timestamp loopEnd,
 	                      DisposeAfterUse::Flag disposeAfterUse = DisposeAfterUse::YES);
-	~SubLoopingAudioStream();
 
 	int readBuffer(int16 *buffer, const int numSamples);
 	bool endOfData() const { return _done; }
@@ -254,8 +252,7 @@ public:
 	bool isStereo() const { return _parent->isStereo(); }
 	int getRate() const { return _parent->getRate(); }
 private:
-	SeekableAudioStream *_parent;
-	DisposeAfterUse::Flag _disposeAfterUse;
+	Common::DisposablePtr<SeekableAudioStream> _parent;
 
 	uint _loops;
 	Timestamp _pos;
@@ -283,7 +280,6 @@ public:
 	 * @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);
 
@@ -297,8 +293,7 @@ public:
 
 	Timestamp getLength() const { return _length; }
 private:
-	SeekableAudioStream *_parent;
-	DisposeAfterUse::Flag _disposeAfterUse;
+	Common::DisposablePtr<SeekableAudioStream> _parent;
 
 	const Timestamp _start;
 	const Timestamp _length;
diff --git a/audio/decoders/aac.cpp b/audio/decoders/aac.cpp
index 874062a702..7700bb3215 100644
--- a/audio/decoders/aac.cpp
+++ b/audio/decoders/aac.cpp
@@ -8,19 +8,16 @@
  * modify it under the terms of the GNU General Public License
  * as published by the Free Software Foundation; either version 2
  * of the License, or (at your option) any later version.
-
+ *
  * This program is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
  * GNU General Public License for more details.
-
+ *
  * You should have received a copy of the GNU General Public License
  * along with this program; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
  *
- * $URL$
- * $Id$
- *
  */
 
 #include "audio/decoders/aac.h"
@@ -28,74 +25,34 @@
 #ifdef USE_FAAD
 
 #include "common/debug.h"
+#include "common/memstream.h"
 #include "common/stream.h"
 #include "common/textconsole.h"
 #include "common/util.h"
 
 #include "audio/audiostream.h"
+#include "audio/decoders/codec.h"
+#include "audio/decoders/raw.h"
 
 #include <neaacdec.h>
 
 namespace Audio {
 
-class AACStream : public AudioStream {
+class AACDecoder : public Codec {
 public:
-	AACStream(Common::SeekableReadStream *stream,
-	          DisposeAfterUse::Flag disposeStream,
-	          Common::SeekableReadStream *extraData,
+	AACDecoder(Common::SeekableReadStream *extraData,
 	          DisposeAfterUse::Flag disposeExtraData);
-	~AACStream();
-
-	int readBuffer(int16 *buffer, const int numSamples);
+	~AACDecoder();
 
-	bool endOfData() const { return _inBufferPos >= _inBufferSize && !_remainingSamples; }
-	bool isStereo() const { return _channels == 2; }
-	int getRate() const { return _rate; }
+	AudioStream *decodeFrame(Common::SeekableReadStream &stream);
 
 private:
 	NeAACDecHandle _handle;
 	byte _channels;
 	unsigned long _rate;
-
-	byte *_inBuffer;
-	uint32 _inBufferSize;
-	uint32 _inBufferPos;
-
-	int16 *_remainingSamples;
-	uint32 _remainingSamplesSize;
-	uint32 _remainingSamplesPos;
-
-	void init(Common::SeekableReadStream *extraData);
 };
 
-AACStream::AACStream(Common::SeekableReadStream *stream, DisposeAfterUse::Flag disposeStream,
-		Common::SeekableReadStream *extraData, DisposeAfterUse::Flag disposeExtraData) {
-
-	_remainingSamples = 0;
-	_inBufferPos = 0;
-
-	init(extraData);
-
-	// Copy all the data to a pointer so it can be passed through
-	// (At least MPEG-4 chunks shouldn't be large)
-	_inBufferSize = stream->size();
-	_inBuffer = new byte[_inBufferSize];
-	stream->read(_inBuffer, _inBufferSize);
-
-	if (disposeStream == DisposeAfterUse::YES)
-		delete stream;
-
-	if (disposeExtraData == DisposeAfterUse::YES)
-		delete extraData;
-}
-
-AACStream::~AACStream() {
-	NeAACDecClose(_handle);
-	delete[] _inBuffer;
-	delete[] _remainingSamples;
-}
-
-void AACStream::init(Common::SeekableReadStream *extraData) {
+AACDecoder::AACDecoder(Common::SeekableReadStream *extraData, DisposeAfterUse::Flag disposeExtraData) {
 	// Open the library
 	_handle = NeAACDecOpen();
 
@@ -117,59 +74,55 @@ void AACStream::init(Common::SeekableReadStream *extraData) {
 
 	if (err < 0)
 		error("Could not initialize AAC decoder: %s", NeAACDecGetErrorMessage(err));
-}
 
-int AACStream::readBuffer(int16 *buffer, const int numSamples) {
-	int samples = 0;
-
-	assert((numSamples % _channels) == 0);
+	if (disposeExtraData == DisposeAfterUse::YES)
+		delete extraData;
+}
 
-	// Dip into our remaining samples pool if it's available
-	if (_remainingSamples) {
-		samples = MIN<int>(numSamples, _remainingSamplesSize - _remainingSamplesPos);
+AACDecoder::~AACDecoder() {
+	NeAACDecClose(_handle);
+}
 
-		memcpy(buffer, _remainingSamples + _remainingSamplesPos, samples * 2);
-		_remainingSamplesPos += samples;
+AudioStream *AACDecoder::decodeFrame(Common::SeekableReadStream &stream) {
+	// read everything into a buffer
+	uint32 inBufferPos = 0;
+	uint32 inBufferSize = stream.size();
+	byte *inBuffer = new byte[inBufferSize];
+	stream.read(inBuffer, inBufferSize);
 
-		if (_remainingSamplesPos == _remainingSamplesSize) {
-			delete[] _remainingSamples;
-			_remainingSamples = 0;
-		}
-	}
+	QueuingAudioStream *audioStream = makeQueuingAudioStream(_rate, _channels == 2);
 
 	// Decode until we have enough samples (or there's no more left)
-	while (samples < numSamples && !endOfData()) {
+	while (inBufferPos < inBufferSize) {
 		NeAACDecFrameInfo frameInfo;
-		uint16 *decodedSamples = (uint16 *)NeAACDecDecode(_handle, &frameInfo, _inBuffer + _inBufferPos, _inBufferSize - _inBufferPos);
+		void *decodedSamples = NeAACDecDecode(_handle, &frameInfo, inBuffer + inBufferPos, inBufferSize - inBufferPos);
 
 		if (frameInfo.error != 0)
 			error("Failed to decode AAC frame: %s", NeAACDecGetErrorMessage(frameInfo.error));
 
-		int decodedSampleSize = frameInfo.samples;
-		int copySamples = (decodedSampleSize > (numSamples - samples)) ? (numSamples - samples) : decodedSampleSize;
+		byte *buffer = (byte *)malloc(frameInfo.samples * 2);
+		memcpy(buffer, decodedSamples, frameInfo.samples * 2);
+
+		byte flags = FLAG_16BITS;
 
-		memcpy(buffer + samples, decodedSamples, copySamples * 2);
-		samples += copySamples;
+		if (_channels == 2)
+			flags |= FLAG_STEREO;
 
-		// Copy leftover samples for use in a later readBuffer() call
-		if (copySamples != decodedSampleSize) {
-			_remainingSamplesSize = decodedSampleSize - copySamples;
-			_remainingSamples = new int16[_remainingSamplesSize];
-			_remainingSamplesPos = 0;
-			memcpy(_remainingSamples, decodedSamples + copySamples, _remainingSamplesSize * 2);
-		}
+#ifdef SCUMM_LITTLE_ENDIAN
+		flags |= FLAG_LITTLE_ENDIAN;
+#endif
 
-		_inBufferPos += frameInfo.bytesconsumed;
+		audioStream->queueBuffer(buffer, frameInfo.samples * 2, DisposeAfterUse::YES, flags);
+
+		inBufferPos += frameInfo.bytesconsumed;
 	}
 
-	return samples;
+	return audioStream;
 }
 
 // Factory function
-AudioStream *makeAACStream(Common::SeekableReadStream *stream, DisposeAfterUse::Flag disposeStream,
-		Common::SeekableReadStream *extraData, DisposeAfterUse::Flag disposeExtraData) {
-
-	return new AACStream(stream, disposeStream, extraData, disposeExtraData);
+Codec *makeAACDecoder(Common::SeekableReadStream *extraData, DisposeAfterUse::Flag disposeExtraData) {
+	return new AACDecoder(extraData, disposeExtraData);
 }
 
 } // End of namespace Audio
diff --git a/audio/decoders/aac.h b/audio/decoders/aac.h
index efcbcc6f42..68e322c844 100644
--- a/audio/decoders/aac.h
+++ b/audio/decoders/aac.h
@@ -8,19 +8,16 @@
  * 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.
  *
- * $URL$
- * $Id$
- *
  */
 
 /**
@@ -43,23 +40,19 @@ namespace Common {
 
 namespace Audio {
 
-class AudioStream;
+class Codec;
 
 /**
- * Create a new AudioStream from the AAC data of an MPEG-4 file in the given stream.
+ * Create a new Codec for decoding AAC data of an MPEG-4 file in the given stream.
  *
  * @note This should *only* be called by our QuickTime/MPEG-4 decoder since it relies
  *       on the MPEG-4 extra data. If you want to decode a file using AAC, go use
  *       makeQuickTimeStream() instead!
- * @param stream            the SeekableReadStream from which to read the AAC data
- * @param disposeStream     whether to delete the stream after use
  * @param extraData         the SeekableReadStream from which to read the AAC extra data
  * @param disposeExtraData  whether to delete the extra data stream after use
- * @return  a new AudioStream, or NULL, if an error occurred
+ * @return  a new Codec, or NULL, if an error occurred
  */
-AudioStream *makeAACStream(
-	Common::SeekableReadStream *stream,
-	DisposeAfterUse::Flag disposeStream,
+Codec *makeAACDecoder(
 	Common::SeekableReadStream *extraData,
 	DisposeAfterUse::Flag disposeExtraData = DisposeAfterUse::NO);
 
diff --git a/audio/decoders/adpcm.cpp b/audio/decoders/adpcm.cpp
index f75196c882..535652a0b3 100644
--- a/audio/decoders/adpcm.cpp
+++ b/audio/decoders/adpcm.cpp
@@ -41,8 +41,7 @@ namespace Audio {
 //   <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),
+	: _stream(stream, disposeAfterUse),
 		_startpos(stream->pos()),
 		_endpos(_startpos + size),
 		_channels(channels),
@@ -52,11 +51,6 @@ ADPCMStream::ADPCMStream(Common::SeekableReadStream *stream, DisposeAfterUse::Fl
 	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
@@ -234,7 +228,7 @@ int MSIma_ADPCMStream::readBuffer(int16 *buffer, const int numSamples) {
 
 		while (samples < numSamples && _samplesLeft[0] != 0) {
 			for (int i = 0; i < _channels; i++) {
-				buffer[samples] = _buffer[i][8 - _samplesLeft[i]];
+				buffer[samples + i] = _buffer[i][8 - _samplesLeft[i]];
 				_samplesLeft[i]--;
 			}
 
diff --git a/audio/decoders/adpcm_intern.h b/audio/decoders/adpcm_intern.h
index c9f894fb84..38514d7fca 100644
--- a/audio/decoders/adpcm_intern.h
+++ b/audio/decoders/adpcm_intern.h
@@ -33,6 +33,7 @@
 
 #include "audio/audiostream.h"
 #include "common/endian.h"
+#include "common/ptr.h"
 #include "common/stream.h"
 #include "common/textconsole.h"
 
@@ -41,8 +42,7 @@ namespace Audio {
 
 class ADPCMStream : public RewindableAudioStream {
 protected:
-	Common::SeekableReadStream *_stream;
-	const DisposeAfterUse::Flag _disposeAfterUse;
+	Common::DisposablePtr<Common::SeekableReadStream> _stream;
 	const int32 _startpos;
 	const int32 _endpos;
 	const int _channels;
@@ -62,7 +62,6 @@ protected:
 
 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; }
diff --git a/audio/decoders/codec.h b/audio/decoders/codec.h
new file mode 100644
index 0000000000..93b6878dee
--- /dev/null
+++ b/audio/decoders/codec.h
@@ -0,0 +1,44 @@
+/* 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.
+ *
+ */
+
+#ifndef AUDIO_DECODERS_CODEC_H
+#define AUDIO_DECODERS_CODEC_H
+
+namespace Common {
+	class SeekableReadStream;
+}
+
+namespace Audio {
+
+class AudioStream;
+
+class Codec {
+public:
+	Codec() {}
+	virtual ~Codec() {}
+
+	virtual AudioStream *decodeFrame(Common::SeekableReadStream &data) = 0;
+};
+
+} // End of namespace Audio
+
+#endif
diff --git a/audio/decoders/mp3.cpp b/audio/decoders/mp3.cpp
index 8d7f006ec7..00669945c2 100644
--- a/audio/decoders/mp3.cpp
+++ b/audio/decoders/mp3.cpp
@@ -25,6 +25,7 @@
 #ifdef USE_MAD
 
 #include "common/debug.h"
+#include "common/ptr.h"
 #include "common/stream.h"
 #include "common/textconsole.h"
 #include "common/util.h"
@@ -52,8 +53,7 @@ protected:
 		MP3_STATE_EOS		// end of data reached (may need to loop)
 	};
 
-	Common::SeekableReadStream *_inStream;
-	DisposeAfterUse::Flag _disposeAfterUse;
+	Common::DisposablePtr<Common::SeekableReadStream> _inStream;
 
 	uint _posInFrame;
 	State _state;
@@ -95,8 +95,7 @@ protected:
 };
 
 MP3Stream::MP3Stream(Common::SeekableReadStream *inStream, DisposeAfterUse::Flag dispose) :
-	_inStream(inStream),
-	_disposeAfterUse(dispose),
+	_inStream(inStream, dispose),
 	_posInFrame(0),
 	_state(MP3_STATE_INIT),
 	_length(0, 1000),
@@ -134,9 +133,6 @@ MP3Stream::MP3Stream(Common::SeekableReadStream *inStream, DisposeAfterUse::Flag
 
 MP3Stream::~MP3Stream() {
 	deinitStream();
-
-	if (_disposeAfterUse == DisposeAfterUse::YES)
-		delete _inStream;
 }
 
 void MP3Stream::decodeMP3Data() {
diff --git a/audio/decoders/qdm2.cpp b/audio/decoders/qdm2.cpp
index a178c363b5..19b30217e9 100644
--- a/audio/decoders/qdm2.cpp
+++ b/audio/decoders/qdm2.cpp
@@ -28,10 +28,13 @@
 #ifdef AUDIO_QDM2_H
 
 #include "audio/audiostream.h"
+#include "audio/decoders/codec.h"
 #include "audio/decoders/qdm2data.h"
+#include "audio/decoders/raw.h"
 
 #include "common/array.h"
 #include "common/debug.h"
+#include "common/math.h"
 #include "common/stream.h"
 #include "common/textconsole.h"
 
@@ -150,19 +153,14 @@ struct RDFTContext {
 	FFTContext fft;
 };
 
-class QDM2Stream : public AudioStream {
+class QDM2Stream : public Codec {
 public:
-	QDM2Stream(Common::SeekableReadStream *stream, Common::SeekableReadStream *extraData);
+	QDM2Stream(Common::SeekableReadStream *extraData, DisposeAfterUse::Flag disposeExtraData);
 	~QDM2Stream();
 
-	bool isStereo() const { return _channels == 2; }
-	bool endOfData() const { return _stream->pos() >= _stream->size() && _outputSamples.size() == 0 && _subPacket == 0; }
-	int getRate() const { return _sampleRate; }
-	int readBuffer(int16 *buffer, const int numSamples);
+	AudioStream *decodeFrame(Common::SeekableReadStream &stream);
 
 private:
-	Common::SeekableReadStream *_stream;
-
 	// Parameters from codec header, do not change during playback
 	uint8 _channels;
 	uint16 _sampleRate;
@@ -204,7 +202,6 @@ private:
 	// I/O data
 	uint8 *_compressedData;
 	float _outputBuffer[1024];
-	Common::Array<int16> _outputSamples;
 
 	// Synthesis filter
 	int16 ff_mpa_synth_window[512];
@@ -285,7 +282,7 @@ private:
 	void qdm2_fft_tone_synthesizer(uint8 sub_packet);
 	void qdm2_calculate_fft(int channel);
 	void qdm2_synthesis_filter(uint8 index);
-	int qdm2_decodeFrame(Common::SeekableReadStream *in);
+	bool qdm2_decodeFrame(Common::SeekableReadStream &in, QueuingAudioStream *audioStream);
 };
 
 // Fix compilation for non C99-compliant compilers, like MSVC
@@ -293,21 +290,6 @@ private:
 typedef signed long long int int64_t;
 #endif
 
-// Integer log2 function. This is much faster than invoking
-// double precision C99 log2 math functions or equivalent, since
-// this is only used to determine maximum number of bits needed
-// i.e. only non-fractional part is needed. Also, the double
-// version is incorrect for exact cases due to floating point
-// rounding errors.
-static inline int scummvm_log2(int n) {
-	int ret = -1;
-	while(n != 0) {
-		n /= 2;
-		ret++;
-	}
-	return ret;
-}
-
 #define QDM2_LIST_ADD(list, size, packet) \
 	do { \
 		if (size > 0) \
@@ -1711,7 +1693,7 @@ void QDM2Stream::initVlc(void) {
 	}
 }
 
-QDM2Stream::QDM2Stream(Common::SeekableReadStream *stream, Common::SeekableReadStream *extraData) {
+QDM2Stream::QDM2Stream(Common::SeekableReadStream *extraData, DisposeAfterUse::Flag disposeExtraData) {
 	uint32 tmp;
 	int32 tmp_s;
 	int tmp_val;
@@ -1719,7 +1701,6 @@ QDM2Stream::QDM2Stream(Common::SeekableReadStream *stream, Common::SeekableReadS
 
 	debug(1, "QDM2Stream::QDM2Stream() Call");
 
-	_stream = stream;
 	_compressedData = NULL;
 	_subPacket = 0;
 	_superBlockStart = 0;
@@ -1841,11 +1822,11 @@ QDM2Stream::QDM2Stream(Common::SeekableReadStream *stream, Common::SeekableReadS
 			warning("QDM2Stream::QDM2Stream() u4 field not 0");
 	}
 
-	_fftOrder = scummvm_log2(_frameSize) + 1;
+	_fftOrder = Common::intLog2(_frameSize) + 1;
 	_fftFrameSize = 2 * _frameSize; // complex has two floats
 
 	// something like max decodable tones
-	_groupOrder = scummvm_log2(_blockSize) + 1;
+	_groupOrder = Common::intLog2(_blockSize) + 1;
 	_sFrameSize = _blockSize / 16; // 16 iterations per super block
 
 	_subSampling = _fftOrder - 7;
@@ -1906,11 +1887,13 @@ QDM2Stream::QDM2Stream(Common::SeekableReadStream *stream, Common::SeekableReadS
 	initNoiseSamples();
 
 	_compressedData = new uint8[_packetSize];
+
+	if (disposeExtraData == DisposeAfterUse::YES)
+		delete extraData;
 }
 
 QDM2Stream::~QDM2Stream() {
 	delete[] _compressedData;
-	delete _stream;
 }
 
 static int qdm2_get_vlc(GetBitContext *gb, VLC *vlc, int flag, int depth) {
@@ -3158,30 +3141,30 @@ void QDM2Stream::qdm2_synthesis_filter(uint8 index)
 			_outputBuffer[_channels * i + ch] += (float)(samples[_channels * sub_sampling * i + ch] >> (sizeof(int16)*8-16));
 }
 
-int QDM2Stream::qdm2_decodeFrame(Common::SeekableReadStream *in) {
-	debug(1, "QDM2Stream::qdm2_decodeFrame in->pos(): %d in->size(): %d", in->pos(), in->size());
+bool QDM2Stream::qdm2_decodeFrame(Common::SeekableReadStream &in, QueuingAudioStream *audioStream) {
+	debug(1, "QDM2Stream::qdm2_decodeFrame in.pos(): %d in.size(): %d", in.pos(), in.size());
 	int ch, i;
 	const int frame_size = (_sFrameSize * _channels);
 
 	// If we're in any packet but the first, seek back to the first
 	if (_subPacket == 0)
-		_superBlockStart = in->pos();
+		_superBlockStart = in.pos();
 	else
-		in->seek(_superBlockStart);
+		in.seek(_superBlockStart);
 
 	// select input buffer
-	if (in->eos() || in->pos() >= in->size()) {
+	if (in.eos() || in.pos() >= in.size()) {
 		debug(1, "QDM2Stream::qdm2_decodeFrame End of Input Stream");
-		return 0;
+		return false;
 	}
 
-	if ((in->size() - in->pos()) < _packetSize) {
-		debug(1, "QDM2Stream::qdm2_decodeFrame Insufficient Packet Data in Input Stream Found: %d Need: %d", in->size() - in->pos(), _packetSize);
-		return 0;
+	if ((in.size() - in.pos()) < _packetSize) {
+		debug(1, "QDM2Stream::qdm2_decodeFrame Insufficient Packet Data in Input Stream Found: %d Need: %d", in.size() - in.pos(), _packetSize);
+		return false;
 	}
 
-	if (!in->eos()) {
-		in->read(_compressedData, _packetSize);
+	if (!in.eos()) {
+		in.read(_compressedData, _packetSize);
 		debug(1, "QDM2Stream::qdm2_decodeFrame constructed input data");
 	}
 
@@ -3190,7 +3173,7 @@ int QDM2Stream::qdm2_decodeFrame(Common::SeekableReadStream *in) {
 	memset(&_outputBuffer[frame_size], 0, frame_size * sizeof(float));
 	debug(1, "QDM2Stream::qdm2_decodeFrame cleared outputBuffer");
 
-	if (!in->eos()) {
+	if (!in.eos()) {
 		// decode block of QDM2 compressed data
 		debug(1, "QDM2Stream::qdm2_decodeFrame decode block of QDM2 compressed data");
 		if (_subPacket == 0) {
@@ -3218,7 +3201,7 @@ int QDM2Stream::qdm2_decodeFrame(Common::SeekableReadStream *in) {
 
 			if (!_hasErrors && _subPacketListC[0].packet != NULL) {
 				error("QDM2 : has errors, and C list is not empty");
-				return 0;
+				return false;
 			}
 		}
 
@@ -3236,6 +3219,12 @@ int QDM2Stream::qdm2_decodeFrame(Common::SeekableReadStream *in) {
 		debug(1, "QDM2Stream::qdm2_decodeFrame clip and convert output float[] to 16bit signed samples");
 	}
 
+	if (frame_size == 0)
+		return false;
+
+	// Prepare a buffer for queuing
+	uint16 *outputBuffer = (uint16 *)malloc(frame_size * 2);
+
 	for (i = 0; i < frame_size; i++) {
 		int value = (int)_outputBuffer[i];
 
@@ -3244,34 +3233,35 @@ int QDM2Stream::qdm2_decodeFrame(Common::SeekableReadStream *in) {
 		else if (value < -SOFTCLIP_THRESHOLD)
 			value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -_softclipTable[-value - SOFTCLIP_THRESHOLD];
 
-		_outputSamples.push_back(value);
+		outputBuffer[i] = value;
 	}
-	return frame_size;
-}
 
-int QDM2Stream::readBuffer(int16 *buffer, const int numSamples) {
-	debug(1, "QDM2Stream::readBuffer numSamples: %d", numSamples);
-	int32 decodedSamples = _outputSamples.size();
-	int32 i;
+	// Queue the translated buffer to our stream
+	byte flags = FLAG_16BITS;
 
-	while (decodedSamples < numSamples) {
-		i = qdm2_decodeFrame(_stream);
-		if (i == 0)
-			break; // Out Of Decode Frames...
-		decodedSamples += i;
-	}
+	if (_channels == 2)
+		flags |= FLAG_STEREO;
+
+#ifdef SCUMM_LITTLE_ENDIAN
+	flags |= FLAG_LITTLE_ENDIAN;
+#endif
+
+	audioStream->queueBuffer((byte *)outputBuffer, frame_size * 2, DisposeAfterUse::YES, flags);
+
+	return true;
+}
 
-	if (decodedSamples > numSamples)
-		decodedSamples = numSamples;
+AudioStream *QDM2Stream::decodeFrame(Common::SeekableReadStream &stream) {
+	QueuingAudioStream *audioStream = makeQueuingAudioStream(_sampleRate, _channels == 2);
 
-	for (i = 0; i < decodedSamples; i++)
-		buffer[i] = _outputSamples.remove_at(0);
+	while (qdm2_decodeFrame(stream, audioStream))
+		;
 
-	return decodedSamples;
+	return audioStream;
 }
 
-AudioStream *makeQDM2Stream(Common::SeekableReadStream *stream, Common::SeekableReadStream *extraData) {
-	return new QDM2Stream(stream, extraData);
+Codec *makeQDM2Decoder(Common::SeekableReadStream *extraData, DisposeAfterUse::Flag disposeExtraData) {
+	return new QDM2Stream(extraData, disposeExtraData);
 }
 
 } // End of namespace Audio
diff --git a/audio/decoders/qdm2.h b/audio/decoders/qdm2.h
index c0ec647bfd..f0793e3c1e 100644
--- a/audio/decoders/qdm2.h
+++ b/audio/decoders/qdm2.h
@@ -26,22 +26,25 @@
 #ifndef AUDIO_QDM2_H
 #define AUDIO_QDM2_H
 
+#include "common/types.h"
+
 namespace Common {
 class SeekableReadStream;
 }
 
 namespace Audio {
 
-class AudioStream;
+class Codec;
 
 /**
- * Create a new AudioStream from the QDM2 data in the given stream.
+ * Create a new Codec from the QDM2 data in the given stream.
  *
- * @param stream       the SeekableReadStream from which to read the FLAC data
- * @param extraData    the QuickTime extra data stream
- * @return   a new AudioStream, or NULL, if an error occurred
+ * @param extraData           the QuickTime extra data stream
+ * @param disposeExtraData    the QuickTime extra data stream
+ * @return   a new Codec, or NULL, if an error occurred
  */
-AudioStream *makeQDM2Stream(Common::SeekableReadStream *stream, Common::SeekableReadStream *extraData);
+Codec *makeQDM2Decoder(Common::SeekableReadStream *extraData,
+                       DisposeAfterUse::Flag disposeExtraData = DisposeAfterUse::NO);
 
 } // End of namespace Audio
 
diff --git a/audio/decoders/quicktime.cpp b/audio/decoders/quicktime.cpp
index 0ad2821cd5..8cf0305e88 100644
--- a/audio/decoders/quicktime.cpp
+++ b/audio/decoders/quicktime.cpp
@@ -8,19 +8,16 @@
  * 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"
@@ -30,6 +27,7 @@
 #include "common/textconsole.h"
 
 #include "audio/audiostream.h"
+#include "audio/decoders/codec.h"
 #include "audio/decoders/quicktime.h"
 #include "audio/decoders/quicktime_intern.h"
 
@@ -86,6 +84,9 @@ void QuickTimeAudioDecoder::init() {
 			// Make sure the bits per sample transfers to the sample size
 			if (entry->getCodecTag() == MKTAG('r', 'a', 'w', ' ') || entry->getCodecTag() == MKTAG('t', 'w', 'o', 's'))
 				_tracks[_audioTrackIndex]->sampleSize = (entry->_bitsPerSample / 8) * entry->_channels;
+
+			// Initialize the codec (if necessary)
+			entry->initCodec();
 		}
 	}
 }
@@ -217,6 +218,9 @@ void QuickTimeAudioDecoder::setAudioStreamPos(const Timestamp &where) {
 	Audio::QuickTimeAudioDecoder::AudioSampleDesc *entry = (Audio::QuickTimeAudioDecoder::AudioSampleDesc *)_tracks[_audioTrackIndex]->sampleDescs[0];
 	_audStream = Audio::makeQueuingAudioStream(entry->_sampleRate, entry->_channels == 2);
 
+	// Reinitialize the codec
+	entry->initCodec();
+
 	// First, we need to track down what audio sample we need
 	Audio::Timestamp curAudioTime = where.convertToFramerate(_tracks[_audioTrackIndex]->timeScale);
 	uint32 sample = curAudioTime.totalNumberOfFrames();
@@ -266,6 +270,11 @@ QuickTimeAudioDecoder::AudioSampleDesc::AudioSampleDesc(Common::QuickTimeParser:
 	_samplesPerFrame = 0;
 	_bytesPerFrame = 0;
 	_bitsPerSample = 0;
+	_codec = 0;
+}
+
+QuickTimeAudioDecoder::AudioSampleDesc::~AudioSampleDesc() {
+	delete _codec;
 }
 
 bool QuickTimeAudioDecoder::AudioSampleDesc::isAudioCodecSupported() const {
@@ -313,7 +322,12 @@ AudioStream *QuickTimeAudioDecoder::AudioSampleDesc::createAudioStream(Common::S
 	if (!stream)
 		return 0;
 
-	if (_codecTag == MKTAG('t', 'w', 'o', 's') || _codecTag == MKTAG('r', 'a', 'w', ' ')) {
+	if (_codec) {
+		// If we've loaded a codec, make sure we use first
+		AudioStream *audioStream = _codec->decodeFrame(*stream);
+		delete stream;
+		return audioStream;
+	} else if (_codecTag == MKTAG('t', 'w', 'o', 's') || _codecTag == MKTAG('r', 'a', 'w', ' ')) {
 		// Fortunately, most of the audio used in Myst videos is raw...
 		uint16 flags = 0;
 		if (_codecTag == MKTAG('r', 'a', 'w', ' '))
@@ -330,24 +344,32 @@ AudioStream *QuickTimeAudioDecoder::AudioSampleDesc::createAudioStream(Common::S
 	} else if (_codecTag == MKTAG('i', 'm', 'a', '4')) {
 		// Riven uses this codec (as do some Myst ME videos)
 		return makeADPCMStream(stream, DisposeAfterUse::YES, stream->size(), kADPCMApple, _sampleRate, _channels, 34);
-	} else if (_codecTag == MKTAG('m', 'p', '4', 'a')) {
-		// The 7th Guest iOS uses an MPEG-4 codec
-#ifdef USE_FAAD
-		if (_parentTrack->objectTypeMP4 == 0x40)
-			return makeAACStream(stream, DisposeAfterUse::YES, _parentTrack->extraData);
-#endif
-#ifdef AUDIO_QDM2_H
-	} else if (_codecTag == MKTAG('Q', 'D', 'M', '2')) {
-		// Myst ME uses this codec for many videos
-		return makeQDM2Stream(stream, _parentTrack->extraData);
-#endif
 	}
 
 	error("Unsupported audio codec");
-
 	return NULL;
 }
 
+void QuickTimeAudioDecoder::AudioSampleDesc::initCodec() {
+	delete _codec; _codec = 0;
+
+	switch (_codecTag) {
+	case MKTAG('Q', 'D', 'M', '2'):
+#ifdef AUDIO_QDM2_H
+		_codec = makeQDM2Decoder(_parentTrack->extraData);
+#endif
+		break;
+	case MKTAG('m', 'p', '4', 'a'):
+#ifdef USE_FAAD
+		if (_parentTrack->objectTypeMP4 == 0x40)
+			_codec = makeAACDecoder(_parentTrack->extraData);
+#endif
+		break;
+	default:
+		break;
+	}
+}
+
 /**
  * A wrapper around QuickTimeAudioDecoder that implements the RewindableAudioStream API
  */
diff --git a/audio/decoders/quicktime.h b/audio/decoders/quicktime.h
index 9f6c6c20e0..4dd1a57710 100644
--- a/audio/decoders/quicktime.h
+++ b/audio/decoders/quicktime.h
@@ -8,19 +8,16 @@
  * 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.
  *
- * $URL$
- * $Id$
- *
  */
 
 /**
diff --git a/audio/decoders/quicktime_intern.h b/audio/decoders/quicktime_intern.h
index f288d5604b..e31a1d3872 100644
--- a/audio/decoders/quicktime_intern.h
+++ b/audio/decoders/quicktime_intern.h
@@ -8,19 +8,16 @@
  * 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.
  *
- * $URL$
- * $Id$
- *
  */
 
 /**
@@ -45,6 +42,7 @@ namespace Common {
 namespace Audio {
 
 class AudioStream;
+class Codec;
 class QueuingAudioStream;
 
 class QuickTimeAudioDecoder : public Common::QuickTimeParser {
@@ -68,10 +66,12 @@ protected:
 	class AudioSampleDesc : public Common::QuickTimeParser::SampleDesc {
 	public:
 		AudioSampleDesc(Common::QuickTimeParser::Track *parentTrack, uint32 codecTag);
+		~AudioSampleDesc();
 
 		bool isAudioCodecSupported() const;
 		uint32 getAudioChunkSampleCount(uint chunk) const;
 		AudioStream *createAudioStream(Common::SeekableReadStream *stream) const;
+		void initCodec();
 
 		// TODO: Make private in the long run
 		uint16 _bitsPerSample;
@@ -79,6 +79,8 @@ protected:
 		uint32 _sampleRate;
 		uint32 _samplesPerFrame;
 		uint32 _bytesPerFrame;
+
+		Codec *_codec;
 	};
 
 	// Common::QuickTimeParser API
diff --git a/audio/decoders/raw.cpp b/audio/decoders/raw.cpp
index 4789fd0f36..881b8c1d6a 100644
--- a/audio/decoders/raw.cpp
+++ b/audio/decoders/raw.cpp
@@ -51,7 +51,7 @@ 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) {
+		: _rate(rate), _isStereo(stereo), _playtime(0, rate), _stream(stream, disposeStream), _blocks(blocks), _curBlock(_blocks.begin()), _blockLeft(0), _buffer(0) {
 
 		assert(_blocks.size() > 0);
 
@@ -82,9 +82,6 @@ public:
 	}
 
 	~RawStream() {
-		if (_disposeAfterUse == DisposeAfterUse::YES)
-			delete _stream;
-
 		delete[] _buffer;
 	}
 
@@ -98,15 +95,14 @@ public:
 
 	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
+	const int _rate;                                           ///< Sample rate of stream
+	const bool _isStereo;                                      ///< Whether this is an stereo stream
+	Timestamp _playtime;                                       ///< Calculated total play time
+	Common::DisposablePtr<Common::SeekableReadStream> _stream; ///< Stream to read data from
+	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
diff --git a/audio/decoders/voc.cpp b/audio/decoders/voc.cpp
index 74ea4440a1..f06e7f95f2 100644
--- a/audio/decoders/voc.cpp
+++ b/audio/decoders/voc.cpp
@@ -47,7 +47,7 @@ int getSampleRateFromVOCRate(int vocSR) {
 	}
 }
 
-static byte *loadVOCFromStream(Common::ReadStream &stream, int &size, int &rate, int &loops, int &begin_loop, int &end_loop) {
+byte *loadVOCFromStream(Common::ReadStream &stream, int &size, int &rate) {
 	VocFileHeader fileHeader;
 
 	debug(2, "loadVOCFromStream");
@@ -84,8 +84,6 @@ static byte *loadVOCFromStream(Common::ReadStream &stream, int &size, int &rate,
 	int len;
 	byte *ret_sound = 0;
 	size = 0;
-	begin_loop = 0;
-	end_loop = 0;
 
 	while ((code = stream.readByte())) {
 		len = stream.readByte();
@@ -118,14 +116,16 @@ static byte *loadVOCFromStream(Common::ReadStream &stream, int &size, int &rate,
 			debug(9, "VOC Data Block: %d, %d, %d", rate, packing, len);
 			if (packing == 0) {
 				if (size) {
-					ret_sound = (byte *)realloc(ret_sound, size + len);
+					byte *tmp = (byte *)realloc(ret_sound, size + len);
+					if (!tmp)
+						error("Cannot reallocate memory for VOC Data Block");
+
+					ret_sound = tmp;
 				} 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);
 			}
@@ -140,7 +140,7 @@ static byte *loadVOCFromStream(Common::ReadStream &stream, int &size, int &rate,
 			break;
 		case 6:	// begin of loop
 			assert(len == 2);
-			loops = stream.readUint16LE();
+			stream.readUint16LE();
 			break;
 		case 7:	// end of loop
 			assert(len == 0);
@@ -169,15 +169,9 @@ static byte *loadVOCFromStream(Common::ReadStream &stream, int &size, int &rate,
 	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) {
+int parseVOCFormat(Common::SeekableReadStream& stream, RawStreamBlock* block, int &rate) {
 	VocFileHeader fileHeader;
 	int currentBlock = 0;
 	int size = 0;
@@ -215,8 +209,6 @@ int parseVOCFormat(Common::SeekableReadStream& stream, RawStreamBlock* block, in
 
 	int len;
 	size = 0;
-	begin_loop = 0;
-	end_loop = 0;
 
 	while ((code = stream.readByte())) {
 		len = stream.readByte();
@@ -257,8 +249,6 @@ int parseVOCFormat(Common::SeekableReadStream& stream, RawStreamBlock* block, in
 				stream.seek(len, SEEK_CUR);
 
 				size += len;
-				begin_loop = size;
-				end_loop = size;
 			} else {
 				warning("VOC file packing %d unsupported", packing);
 			}
@@ -273,7 +263,7 @@ int parseVOCFormat(Common::SeekableReadStream& stream, RawStreamBlock* block, in
 			break;
 		case 6:	// begin of loop
 			assert(len == 2);
-			loops = stream.readUint16LE();
+			stream.readUint16LE();
 			break;
 		case 7:	// end of loop
 			assert(len == 0);
@@ -296,33 +286,13 @@ int parseVOCFormat(Common::SeekableReadStream& stream, RawStreamBlock* block, in
 	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 rate;
 
-	int numBlocks = parseVOCFormat(*stream, block, rate, loops, begin_loop, end_loop);
+	int numBlocks = parseVOCFormat(*stream, block, rate);
 
 	SeekableAudioStream *audioStream = 0;
 
@@ -338,47 +308,6 @@ SeekableAudioStream *makeVOCDiskStreamNoLoop(Common::SeekableReadStream *stream,
 
 #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);
diff --git a/audio/decoders/voc.h b/audio/decoders/voc.h
index 8bc6dcf46f..a920eac933 100644
--- a/audio/decoders/voc.h
+++ b/audio/decoders/voc.h
@@ -24,9 +24,7 @@
  * @file
  * Sound decoder used in engines:
  *  - agos
- *  - drascula
  *  - kyra
- *  - made
  *  - saga
  *  - scumm
  *  - touche
@@ -90,16 +88,11 @@ 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.
+ * 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);
+SeekableAudioStream *makeVOCStream(Common::SeekableReadStream *stream, byte flags, DisposeAfterUse::Flag disposeAfterUse = DisposeAfterUse::NO);
 
 } // End of namespace Audio
 
diff --git a/audio/decoders/vorbis.cpp b/audio/decoders/vorbis.cpp
index 2724dd1f02..64cacb4d58 100644
--- a/audio/decoders/vorbis.cpp
+++ b/audio/decoders/vorbis.cpp
@@ -29,6 +29,7 @@
 
 #ifdef USE_VORBIS
 
+#include "common/ptr.h"
 #include "common/stream.h"
 #include "common/textconsole.h"
 #include "common/util.h"
@@ -42,6 +43,7 @@
 #include <tremor/ivorbisfile.h>
 #endif
 #else
+#define OV_EXCLUDE_STATIC_CALLBACKS
 #include <vorbis/vorbisfile.h>
 #endif
 
@@ -88,8 +90,7 @@ static ov_callbacks g_stream_wrap = {
 
 class VorbisStream : public SeekableAudioStream {
 protected:
-	Common::SeekableReadStream *_inStream;
-	DisposeAfterUse::Flag _disposeAfterUse;
+	Common::DisposablePtr<Common::SeekableReadStream> _inStream;
 
 	bool _isStereo;
 	int _rate;
@@ -120,10 +121,9 @@ protected:
 };
 
 VorbisStream::VorbisStream(Common::SeekableReadStream *inStream, DisposeAfterUse::Flag dispose) :
-	_inStream(inStream),
-	_disposeAfterUse(dispose),
+	_inStream(inStream, dispose),
 	_length(0, 1000),
-	_bufferEnd(_buffer + ARRAYSIZE(_buffer)) {
+	_bufferEnd(ARRAYEND(_buffer)) {
 
 	int res = ov_open_callbacks(inStream, &_ovFile, NULL, 0, g_stream_wrap);
 	if (res < 0) {
@@ -149,8 +149,6 @@ VorbisStream::VorbisStream(Common::SeekableReadStream *inStream, DisposeAfterUse
 
 VorbisStream::~VorbisStream() {
 	ov_clear(&_ovFile);
-	if (_disposeAfterUse == DisposeAfterUse::YES)
-		delete _inStream;
 }
 
 int VorbisStream::readBuffer(int16 *buffer, const int numSamples) {
diff --git a/audio/decoders/vag.cpp b/audio/decoders/xa.cpp
index 10ef69708c..818cd2df59 100644
--- a/audio/decoders/vag.cpp
+++ b/audio/decoders/xa.cpp
@@ -20,79 +20,78 @@
  *
  */
 
-#include "audio/decoders/vag.h"
+#include "audio/decoders/xa.h"
 #include "audio/audiostream.h"
 #include "common/stream.h"
 
 namespace Audio {
 
-class VagStream : public Audio::RewindableAudioStream {
+class XAStream : public Audio::RewindableAudioStream {
 public:
-	VagStream(Common::SeekableReadStream *stream, int rate);
-	~VagStream();
+	XAStream(Common::SeekableReadStream *stream, int rate, DisposeAfterUse::Flag disposeAfterUse);
+	~XAStream();
 
 	bool isStereo() const { return false; }
-	bool endOfData() const { return _stream->pos() == _stream->size(); }
+	bool endOfData() const { return _endOfData && _samplesRemaining == 0; }
 	int getRate() const { return _rate; }
 	int readBuffer(int16 *buffer, const int numSamples);
 
 	bool rewind();
 private:
 	Common::SeekableReadStream *_stream;
+	DisposeAfterUse::Flag _disposeAfterUse;
+
+	void seekToPos(uint pos);
 
 	byte _predictor;
 	double _samples[28];
 	byte _samplesRemaining;
 	int _rate;
 	double _s1, _s2;
+	uint _loopPoint;
+	bool _endOfData;
 };
 
-VagStream::VagStream(Common::SeekableReadStream *stream, int rate) : _stream(stream) {
+XAStream::XAStream(Common::SeekableReadStream *stream, int rate, DisposeAfterUse::Flag disposeAfterUse)
+		: _stream(stream), _disposeAfterUse(disposeAfterUse) {
 	_samplesRemaining = 0;
 	_predictor = 0;
 	_s1 = _s2 = 0.0;
 	_rate = rate;
+	_loopPoint = 0;
+	_endOfData = false;
 }
 
 
-VagStream::~VagStream() {
-	delete _stream;
+XAStream::~XAStream() {
+	if (_disposeAfterUse == DisposeAfterUse::YES)
+		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 }
-	};
+static const double s_xaDataTable[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) {
+int XAStream::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;
+	for (int i = 28 - _samplesRemaining; i < 28 && samplesDecoded < numSamples; i++) {
+		_samples[i] = _samples[i] + _s1 * s_xaDataTable[_predictor][0] + _s2 * s_xaDataTable[_predictor][1];
+		_s2 = _s1;
+		_s1 = _samples[i];
+		int16 d = (int) (_samples[i] + 0.5);
+		buffer[samplesDecoded] = d;
+		samplesDecoded++;
+		_samplesRemaining--;
 	}
 
+	if (endOfData())
+		return samplesDecoded;
+
 	while (samplesDecoded < numSamples) {
 		byte i = 0;
 
@@ -100,8 +99,19 @@ int VagStream::readBuffer(int16 *buffer, const int numSamples) {
 		byte shift = _predictor & 0xf;
 		_predictor >>= 4;
 
-		if (_stream->readByte() == 7)
+		byte flags = _stream->readByte();
+		if (flags == 3) {
+			// Loop
+			seekToPos(_loopPoint);
+			continue;
+		} else if (flags == 6) {
+			// Set loop point
+			_loopPoint = _stream->pos() - 2;
+		} else if (flags == 7) {
+			// End of stream
+			_endOfData = true;
 			return samplesDecoded;
+		}
 
 		for (i = 0; i < 28; i += 2) {
 			byte d = _stream->readByte();
@@ -116,7 +126,7 @@ int VagStream::readBuffer(int16 *buffer, const int numSamples) {
 		}
 
 		for (i = 0; i < 28 && samplesDecoded < numSamples; i++) {
-			_samples[i] = _samples[i] + _s1 * s_vagDataTable[_predictor][0] + _s2 * s_vagDataTable[_predictor][1];
+			_samples[i] = _samples[i] + _s1 * s_xaDataTable[_predictor][0] + _s2 * s_xaDataTable[_predictor][1];
 			_s2 = _s1;
 			_s1 = _samples[i];
 			int16 d = (int) (_samples[i] + 0.5);
@@ -124,24 +134,31 @@ int VagStream::readBuffer(int16 *buffer, const int numSamples) {
 			samplesDecoded++;
 		}
 
-		if (i != 27)
+		if (i != 28)
 			_samplesRemaining = 28 - i;
+
+		if (_stream->pos() >= _stream->size())
+			_endOfData = true;
 	}
 
 	return samplesDecoded;
 }
 
-bool VagStream::rewind() {
-	_stream->seek(0);
+bool XAStream::rewind() {
+	seekToPos(0);
+	return true;
+}
+
+void XAStream::seekToPos(uint pos) {
+	_stream->seek(pos);
 	_samplesRemaining = 0;
 	_predictor = 0;
 	_s1 = _s2 = 0.0;
-
-	return true;
+	_endOfData = false;
 }
 
-RewindableAudioStream *makeVagStream(Common::SeekableReadStream *stream, int rate) {
-	return new VagStream(stream, rate);
+RewindableAudioStream *makeXAStream(Common::SeekableReadStream *stream, int rate, DisposeAfterUse::Flag disposeAfterUse) {
+	return new XAStream(stream, rate, disposeAfterUse);
 }
 
-}
+} // End of namespace Audio
diff --git a/audio/decoders/vag.h b/audio/decoders/xa.h
index b80fbdb98f..cf28d8001a 100644
--- a/audio/decoders/vag.h
+++ b/audio/decoders/xa.h
@@ -28,8 +28,10 @@
  * - tinsel (PSX port of the game)
  */
 
-#ifndef SOUND_VAG_H
-#define SOUND_VAG_H
+#ifndef AUDIO_DECODERS_XA_H
+#define AUDIO_DECODERS_XA_H
+
+#include "common/types.h"
 
 namespace Common {
 class SeekableReadStream;
@@ -40,17 +42,19 @@ namespace Audio {
 class RewindableAudioStream;
 
 /**
- * Takes an input stream containing Vag sound data and creates
+ * Takes an input stream containing XA ADPCM sound data and creates
  * an RewindableAudioStream from that.
  *
- * @param stream            the SeekableReadStream from which to read the ADPCM data
+ * @param stream            the SeekableReadStream from which to read the XA ADPCM data
  * @param rate              the sampling rate
+ * @param disposeAfterUse   whether to delete the stream after use.
  * @return   a new RewindableAudioStream, or NULL, if an error occurred
  */
-RewindableAudioStream *makeVagStream(
+RewindableAudioStream *makeXAStream(
 	Common::SeekableReadStream *stream,
-	int rate = 11025);
+	int rate,
+	DisposeAfterUse::Flag disposeAfterUse = DisposeAfterUse::YES);
 
-} // End of namespace Sword1
+} // End of namespace Audio
 
 #endif
diff --git a/audio/fmopl.cpp b/audio/fmopl.cpp
index a24c2a533c..da655643a7 100644
--- a/audio/fmopl.cpp
+++ b/audio/fmopl.cpp
@@ -192,4 +192,3 @@ FM_OPL *makeAdLibOPL(int rate) {
 
 	return opl;
 }
-
diff --git a/audio/fmopl.h b/audio/fmopl.h
index b88325a52e..f62587f557 100644
--- a/audio/fmopl.h
+++ b/audio/fmopl.h
@@ -176,4 +176,3 @@ void YM3812UpdateOne(FM_OPL *OPL, int16 *buffer, int length);
 FM_OPL *makeAdLibOPL(int rate);
 
 #endif
-
diff --git a/audio/mididrv.cpp b/audio/mididrv.cpp
index b71c02f991..f638250dcd 100644
--- a/audio/mididrv.cpp
+++ b/audio/mididrv.cpp
@@ -55,27 +55,30 @@ const byte MidiDriver::_gmToMt32[128] = {
 	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
+static const struct {
+	uint32      type;
+	const char *guio;
+} GUIOMapping[] = {
+	{ MT_PCSPK,		GUIO_MIDIPCSPK, },
+	{ MT_CMS,		GUIO_MIDICMS, },
+	{ MT_PCJR,		GUIO_MIDIPCJR, },
+	{ MT_ADLIB,		GUIO_MIDIADLIB, },
+	{ MT_C64,		GUIO_MIDIC64, },
+	{ MT_AMIGA,	    GUIO_MIDIAMIGA, },
+	{ MT_APPLEIIGS,	GUIO_MIDIAPPLEIIGS, },
+	{ MT_TOWNS,		GUIO_MIDITOWNS, },
+	{ MT_PC98,		GUIO_MIDIPC98, },
+	{ MT_GM,		GUIO_MIDIGM, },
+	{ MT_MT32,		GUIO_MIDIMT32, },
+	{ 0,			0 },
 };
 
-uint32 MidiDriver::musicType2GUIO(uint32 musicType) {
-	uint32 res = 0;
+Common::String MidiDriver::musicType2GUIO(uint32 musicType) {
+	Common::String res = "";
 
-	for (int i = 0; GUIOMapping[i] || GUIOMapping[i + 1]; i += 2) {
-		if (musicType == GUIOMapping[i] || musicType == (uint32)-1)
-			res |= GUIOMapping[i + 1];
+	for (int i = 0; GUIOMapping[i].guio; i++) {
+		if (musicType == GUIOMapping[i].type || musicType == (uint32)-1)
+			res += GUIOMapping[i].guio;
 	}
 
 	return res;
@@ -128,7 +131,8 @@ Common::String MidiDriver::getDeviceString(DeviceHandle handle, DeviceStringType
 
 MidiDriver::DeviceHandle MidiDriver::detectDevice(int flags) {
 	// Query the selected music device (defaults to MT_AUTO device).
-	DeviceHandle hdl = getDeviceHandle(ConfMan.get("music_driver"));
+	Common::String selDevStr = ConfMan.hasKey("music_driver") ? ConfMan.get("music_driver") : Common::String("auto");
+	DeviceHandle hdl = getDeviceHandle(selDevStr.empty() ? Common::String("auto") : selDevStr);
 	DeviceHandle reslt = 0;
 
 	_forceTypeMT32 = false;
@@ -200,8 +204,8 @@ MidiDriver::DeviceHandle MidiDriver::detectDevice(int flags) {
 	if (getMusicType(hdl) == MT_INVALID) {
 		// If the expressly selected driver or device cannot be found (no longer compiled in, turned off, etc.)
 		// we display a warning and continue.
-		failedDevStr = ConfMan.get("music_driver");
-		Common::String warningMsg = Common::String::format(_("The selected audio device '%s' was not found (e.g. might be turned off or disconnected). Attempting to fall back to the next available device..."), failedDevStr.c_str());
+		failedDevStr = selDevStr;
+		Common::String warningMsg = Common::String::format(_("The selected audio device '%s' was not found (e.g. might be turned off or disconnected)."), failedDevStr.c_str()) + " " + _("Attempting to fall back to the next available device...");
 		GUI::MessageDialog dialog(warningMsg);
 		dialog.runModal();
 	}
@@ -213,7 +217,7 @@ MidiDriver::DeviceHandle MidiDriver::detectDevice(int flags) {
 		} else {
 			// If the expressly selected device cannot be used we display a warning and continue.
 			failedDevStr = getDeviceString(hdl, MidiDriver::kDeviceName);
-			Common::String warningMsg = Common::String::format(_("The selected audio device '%s' cannot be used. See log file for more information. Attempting to fall back to the next available device..."), failedDevStr.c_str());
+			Common::String warningMsg = Common::String::format(_("The selected audio device '%s' cannot be used. See log file for more information."), failedDevStr.c_str()) + " " + _("Attempting to fall back to the next available device...");
 			GUI::MessageDialog dialog(warningMsg);
 			dialog.runModal();
 		}
@@ -230,13 +234,15 @@ MidiDriver::DeviceHandle MidiDriver::detectDevice(int flags) {
 			// If a preferred MT32 or GM device has been selected that device gets returned if available.
 			Common::String devStr;
 			if (flags & MDT_PREFER_MT32)
-				devStr = ConfMan.get("mt32_device");
+				devStr = ConfMan.hasKey("mt32_device") ? ConfMan.get("mt32_device") : Common::String("null");
 			else if (flags & MDT_PREFER_GM)
-				devStr = ConfMan.get("gm_device");
+				devStr = ConfMan.hasKey("gm_device") ? ConfMan.get("gm_device") : Common::String("null");
 			else
 				devStr = "auto";
-
-			hdl = getDeviceHandle(devStr);
+			
+			// Default to Null device here, since we also register a default null setting for
+			// the MT32 or GM device in the config manager.
+			hdl = getDeviceHandle(devStr.empty() ? Common::String("null") : devStr);
 			const MusicType type = getMusicType(hdl);
 
 			// If we have a "Don't use GM/MT-32" setting we skip this part and jump
@@ -247,7 +253,7 @@ MidiDriver::DeviceHandle MidiDriver::detectDevice(int flags) {
 					// we display a warning and continue. Don't warn about the missing device if we did already (this becomes relevant if the
 					// missing device is selected as preferred device and also as GM or MT-32 device).
 					if (failedDevStr != devStr) {
-						Common::String warningMsg = Common::String::format(_("The preferred audio device '%s' was not found (e.g. might be turned off or disconnected). Attempting to fall back to the next available device..."), devStr.c_str());
+						Common::String warningMsg = Common::String::format(_("The preferred audio device '%s' was not found (e.g. might be turned off or disconnected)."), devStr.c_str()) + " " + _("Attempting to fall back to the next available device...");
 						GUI::MessageDialog dialog(warningMsg);
 						dialog.runModal();
 					}
@@ -262,7 +268,7 @@ MidiDriver::DeviceHandle MidiDriver::detectDevice(int flags) {
 						// Don't warn about the failing device if we did already (this becomes relevant if the failing
 						// device is selected as preferred device and also as GM or MT-32 device).
 						if (failedDevStr != getDeviceString(hdl, MidiDriver::kDeviceName)) {
-							Common::String warningMsg = Common::String::format(_("The preferred audio device '%s' cannot be used. See log file for more information. Attempting to fall back to the next available device..."), getDeviceString(hdl, MidiDriver::kDeviceName).c_str());
+							Common::String warningMsg = Common::String::format(_("The preferred audio device '%s' cannot be used. See log file for more information."), getDeviceString(hdl, MidiDriver::kDeviceName).c_str()) + " " + _("Attempting to fall back to the next available device...");
 							GUI::MessageDialog dialog(warningMsg);
 							dialog.runModal();
 						}
@@ -409,4 +415,3 @@ void MidiDriver::sendGMReset() {
 	sysEx(resetSysEx, sizeof(resetSysEx));
 	g_system->delayMillis(100);
 }
-
diff --git a/audio/mididrv.h b/audio/mididrv.h
index e3f6461be9..cdf2943f2a 100644
--- a/audio/mididrv.h
+++ b/audio/mididrv.h
@@ -146,7 +146,7 @@ public:
 		kDeviceId
 	};
 
-	static uint32 musicType2GUIO(uint32 musicType);
+	static Common::String musicType2GUIO(uint32 musicType);
 
 	/** Create music driver matching the given device handle, or NULL if there is no match. */
 	static MidiDriver *createMidi(DeviceHandle handle);
diff --git a/audio/mixer.cpp b/audio/mixer.cpp
index 128224ae85..965766170d 100644
--- a/audio/mixer.cpp
+++ b/audio/mixer.cpp
@@ -163,9 +163,8 @@ private:
 	uint32 _pauseStartTime;
 	uint32 _pauseTime;
 
-	DisposeAfterUse::Flag _autofreeStream;
 	RateConverter *_converter;
-	AudioStream *_stream;
+	Common::DisposablePtr<AudioStream> _stream;
 };
 
 #pragma mark -
@@ -492,8 +491,8 @@ 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) {
+      _pauseStartTime(0), _pauseTime(0), _converter(0),
+      _stream(stream, autofreeStream) {
 	assert(mixer);
 	assert(stream);
 
@@ -503,8 +502,6 @@ Channel::Channel(Mixer *mixer, Mixer::SoundType type, AudioStream *stream,
 
 Channel::~Channel() {
 	delete _converter;
-	if (_autofreeStream == DisposeAfterUse::YES)
-		delete _stream;
 }
 
 void Channel::setVolume(const byte volume) {
diff --git a/audio/mods/maxtrax.cpp b/audio/mods/maxtrax.cpp
index 953bb8f8d2..344d678b76 100644
--- a/audio/mods/maxtrax.cpp
+++ b/audio/mods/maxtrax.cpp
@@ -707,8 +707,8 @@ int8 MaxTrax::noteOn(ChannelContext &channel, const byte note, uint16 volume, ui
 	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)
+		VoiceContext *voice = ARRAYEND(_voiceCtx);
+		for (voiceNum = ARRAYSIZE(_voiceCtx); voiceNum-- != 0 && --voice->channel != &channel;)
 			;
 		if (voiceNum < 0)
 			voiceNum = pickvoice((channel.flags & ChannelContext::kFlagRightChannel) != 0 ? 1 : 0, pri);
diff --git a/audio/module.mk b/audio/module.mk
index 46cb9944e1..e3aa0aaa81 100644
--- a/audio/module.mk
+++ b/audio/module.mk
@@ -23,10 +23,10 @@ MODULE_OBJS := \
 	decoders/qdm2.o \
 	decoders/quicktime.o \
 	decoders/raw.o \
-	decoders/vag.o \
 	decoders/voc.o \
 	decoders/vorbis.o \
 	decoders/wave.o \
+	decoders/xa.o \
 	mods/infogrames.o \
 	mods/maxtrax.o \
 	mods/module.o \
diff --git a/audio/mpu401.cpp b/audio/mpu401.cpp
index caad945258..103a3501db 100644
--- a/audio/mpu401.cpp
+++ b/audio/mpu401.cpp
@@ -33,7 +33,10 @@ void MidiChannel_MPU401::init(MidiDriver *owner, byte channel) {
 bool MidiChannel_MPU401::allocate() {
 	if (_allocated)
 		return false;
-	return (_allocated = true);
+
+	_allocated = true;
+
+	return true;
 }
 
 MidiDriver *MidiChannel_MPU401::device() {
@@ -143,6 +146,6 @@ void MidiDriver_MPU401::setTimerCallback(void *timer_param, Common::TimerManager
 			g_system->getTimerManager()->removeTimerProc(_timer_proc);
 		_timer_proc = timer_proc;
 		if (timer_proc)
-			g_system->getTimerManager()->installTimerProc(timer_proc, 10000, timer_param);
+			g_system->getTimerManager()->installTimerProc(timer_proc, 10000, timer_param, "MPU401");
 	}
 }
diff --git a/audio/rate.cpp b/audio/rate.cpp
index 83abd6150b..0fc23a8a54 100644
--- a/audio/rate.cpp
+++ b/audio/rate.cpp
@@ -298,6 +298,9 @@ public:
 			_bufferSize = osamp;
 		}
 
+		if (!_buffer)
+			error("[CopyRateConverter::flow] Cannot allocate memory for temp buffer");
+
 		// Read up to 'osamp' samples into our temporary buffer
 		len = input.readBuffer(_buffer, osamp);
 
diff --git a/audio/rate_arm.cpp b/audio/rate_arm.cpp
index 433a7d3423..4135cdd1af 100644
--- a/audio/rate_arm.cpp
+++ b/audio/rate_arm.cpp
@@ -467,4 +467,3 @@ RateConverter *makeRateConverter(st_rate_t inrate, st_rate_t outrate, bool stere
 }
 
 } // End of namespace Audio
-
diff --git a/audio/softsynth/appleiigs.cpp b/audio/softsynth/appleiigs.cpp
index 6ee70d1202..bbb3f0b005 100644
--- a/audio/softsynth/appleiigs.cpp
+++ b/audio/softsynth/appleiigs.cpp
@@ -51,4 +51,3 @@ MusicDevices AppleIIGSMusicPlugin::getDevices() const {
 //#else
 	REGISTER_PLUGIN_STATIC(APPLEIIGS, PLUGIN_TYPE_MUSIC, AppleIIGSMusicPlugin);
 //#endif
-
diff --git a/audio/softsynth/cms.cpp b/audio/softsynth/cms.cpp
index 67eacd1a41..a675da3f03 100644
--- a/audio/softsynth/cms.cpp
+++ b/audio/softsynth/cms.cpp
@@ -163,19 +163,15 @@ void CMSEmulator::update(int chip, int16 *buffer, int length) {
 	struct SAA1099 *saa = &_saa1099[chip];
 	int j, ch;
 
+	if (chip == 0) {
+		memset(buffer, 0, sizeof(int16)*length*2);
+	}
+
 	/* 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;
@@ -244,8 +240,8 @@ void CMSEmulator::update(int chip, int16 *buffer, int length) {
 			}
 		}
 		/* write sound data to the buffer */
-		buffer[j*2] += output_l / 6;
-		buffer[j*2+1] += output_r / 6;
+		buffer[j*2+0] = CLIP<int>(buffer[j*2+0] + output_l / 6, -32768, 32767);
+		buffer[j*2+1] = CLIP<int>(buffer[j*2+1] + output_r / 6, -32768, 32767);
 	}
 }
 
diff --git a/audio/softsynth/eas.cpp b/audio/softsynth/eas.cpp
index d829e3b39a..ea79b25329 100644
--- a/audio/softsynth/eas.cpp
+++ b/audio/softsynth/eas.cpp
@@ -480,4 +480,3 @@ Common::Error EASMusicPlugin::createInstance(MidiDriver **mididriver, MidiDriver
 //#endif
 
 #endif
-
diff --git a/audio/softsynth/emumidi.h b/audio/softsynth/emumidi.h
index f3d7645f87..f72dad7eaf 100644
--- a/audio/softsynth/emumidi.h
+++ b/audio/softsynth/emumidi.h
@@ -26,8 +26,6 @@
 #include "audio/mididrv.h"
 #include "audio/mixer.h"
 
-#define FIXP_SHIFT 16
-
 class MidiDriver_Emulated : public Audio::AudioStream, public MidiDriver {
 protected:
 	bool _isOpen;
@@ -38,6 +36,10 @@ private:
 	Common::TimerManager::TimerProc _timerProc;
 	void *_timerParam;
 
+	enum {
+		FIXP_SHIFT = 16
+	};
+
 	int _nextTick;
 	int _samplesPerTick;
 
diff --git a/audio/softsynth/fmtowns_pc98/towns_audio.h b/audio/softsynth/fmtowns_pc98/towns_audio.h
index 4af888f009..211133a1fe 100644
--- a/audio/softsynth/fmtowns_pc98/towns_audio.h
+++ b/audio/softsynth/fmtowns_pc98/towns_audio.h
@@ -53,4 +53,3 @@ private:
 };
 
 #endif
-
diff --git a/audio/softsynth/fmtowns_pc98/towns_euphony.h b/audio/softsynth/fmtowns_pc98/towns_euphony.h
index 6b30bfb7f5..bff0e99660 100644
--- a/audio/softsynth/fmtowns_pc98/towns_euphony.h
+++ b/audio/softsynth/fmtowns_pc98/towns_euphony.h
@@ -181,4 +181,3 @@ private:
 };
 
 #endif
-
diff --git a/audio/softsynth/fmtowns_pc98/towns_midi.cpp b/audio/softsynth/fmtowns_pc98/towns_midi.cpp
index e415a0dda5..b8203944c0 100644
--- a/audio/softsynth/fmtowns_pc98/towns_midi.cpp
+++ b/audio/softsynth/fmtowns_pc98/towns_midi.cpp
@@ -464,9 +464,10 @@ int TownsMidiOutputChannel::advanceEffectEnvelope(EffectEnvelope *s, EffectDef *
 		s->currentLevel = t;
 		s->modWheelLast = s->modWheelState;
 		t = getEffectModLevel(t, s->modWheelState);
-		if (t != d->phase)
+		if (t != d->phase) {
 			d->phase = t;
-		retFlags |= 1;
+			retFlags |= 1;
+		}
 	}
 
 	if (--s->stepCounter)
@@ -562,7 +563,7 @@ int TownsMidiOutputChannel::getEffectModLevel(int lvl, int mod) {
 		if (lvl < 0)
 			return -_driver->_operatorLevelTable[((-lvl) << 5) + mod];
 		else
-			return _driver->_operatorLevelTable[((-lvl) << 5) + mod];
+			return _driver->_operatorLevelTable[(lvl << 5) + mod];
 	}
 
 	return 0;
diff --git a/audio/softsynth/fmtowns_pc98/towns_midi.h b/audio/softsynth/fmtowns_pc98/towns_midi.h
index 8c764c55d9..1143dbaa02 100644
--- a/audio/softsynth/fmtowns_pc98/towns_midi.h
+++ b/audio/softsynth/fmtowns_pc98/towns_midi.h
@@ -81,4 +81,3 @@ private:
 };
 
 #endif
-
diff --git a/audio/softsynth/fmtowns_pc98/towns_pc98_driver.cpp b/audio/softsynth/fmtowns_pc98/towns_pc98_driver.cpp
index 001d258873..05a4079442 100644
--- a/audio/softsynth/fmtowns_pc98/towns_pc98_driver.cpp
+++ b/audio/softsynth/fmtowns_pc98/towns_pc98_driver.cpp
@@ -1449,4 +1449,3 @@ const uint8 TownsPC98_AudioDriver::_drvTables[] = {
 };
 
 #undef EUPHONY_FADEOUT_TICKS
-
diff --git a/audio/softsynth/fmtowns_pc98/towns_pc98_driver.h b/audio/softsynth/fmtowns_pc98/towns_pc98_driver.h
index ff58482227..c0009e4957 100644
--- a/audio/softsynth/fmtowns_pc98/towns_pc98_driver.h
+++ b/audio/softsynth/fmtowns_pc98/towns_pc98_driver.h
@@ -115,4 +115,3 @@ private:
 };
 
 #endif
-
diff --git a/audio/softsynth/fmtowns_pc98/towns_pc98_fmsynth.h b/audio/softsynth/fmtowns_pc98/towns_pc98_fmsynth.h
index 4f81fa9a5c..49700be5dc 100644
--- a/audio/softsynth/fmtowns_pc98/towns_pc98_fmsynth.h
+++ b/audio/softsynth/fmtowns_pc98/towns_pc98_fmsynth.h
@@ -185,4 +185,3 @@ private:
 };
 
 #endif
-
diff --git a/audio/softsynth/mt32.cpp b/audio/softsynth/mt32.cpp
index 6703a6f7b7..eabde21296 100644
--- a/audio/softsynth/mt32.cpp
+++ b/audio/softsynth/mt32.cpp
@@ -486,7 +486,7 @@ void MidiDriver_ThreadedMT32::setTimerCallback(void *timer_param, TimerManager::
 			_vm->_timer->removeTimerProc(_timer_proc);
 		_timer_proc = timer_proc;
 		if (timer_proc)
-			_vm->_timer->installTimerProc(timer_proc, getBaseTempo(), timer_param);
+			_vm->_timer->installTimerProc(timer_proc, getBaseTempo(), timer_param, "MT32tempo");
 	}
 }
 
diff --git a/audio/softsynth/mt32/mt32_file.cpp b/audio/softsynth/mt32/mt32_file.cpp
index cdf9fa13f6..643082b086 100644
--- a/audio/softsynth/mt32/mt32_file.cpp
+++ b/audio/softsynth/mt32/mt32_file.cpp
@@ -67,4 +67,3 @@ bool File::writeBit32u(Bit32u out) {
 }
 
 } // End of namespace MT32Emu
-
diff --git a/audio/softsynth/opl/dbopl.cpp b/audio/softsynth/opl/dbopl.cpp
index 2c46cfee75..02c2317b25 100644
--- a/audio/softsynth/opl/dbopl.cpp
+++ b/audio/softsynth/opl/dbopl.cpp
@@ -1,5 +1,5 @@
 /*
- *  Copyright (C) 2002-2010  The DOSBox Team
+ *  Copyright (C) 2002-2011  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
@@ -32,7 +32,7 @@
 	//DUNNO Keyon in 4op, switch to 2op without keyoff.
 */
 
-// Last synch with DOSBox SVN trunk r3556
+// Last synch with DOSBox SVN trunk r3752
 
 #include "dbopl.h"
 
@@ -572,7 +572,7 @@ INLINE Bits Operator::GetWave( Bitu index, Bitu vol ) {
 	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 ) );
+	Bit32u total = ( wave & 0x7fff ) + vol << ( 3 - ENV_EXTRA );
 	Bit32s sig = ExpTable[ total & 0xff ];
 	Bit32u exp = total >> 8;
 	Bit32s neg = wave >> 16;
@@ -1236,7 +1236,7 @@ void Chip::GenerateBlock2( Bitu total, Bit32s* output ) {
 void Chip::GenerateBlock3( Bitu total, Bit32s* output  ) {
 	while ( total > 0 ) {
 		Bit32u samples = ForwardLFO( total );
-		memset(output, 0, sizeof(Bit32s) * 2 * samples);
+		memset(output, 0, sizeof(Bit32s) * samples * 2);
 		int count = 0;
 		for( Channel* ch = chan; ch < chan + 18; ) {
 			count++;
diff --git a/audio/softsynth/opl/dbopl.h b/audio/softsynth/opl/dbopl.h
index 87d1045fab..3dbd98986d 100644
--- a/audio/softsynth/opl/dbopl.h
+++ b/audio/softsynth/opl/dbopl.h
@@ -1,5 +1,5 @@
 /*
- *  Copyright (C) 2002-2010  The DOSBox Team
+ *  Copyright (C) 2002-2011  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
@@ -16,7 +16,7 @@
  *  Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
  */
 
-// Last synch with DOSBox SVN trunk r3556
+// Last synch with DOSBox SVN trunk r3752
 
 #ifndef SOUND_SOFTSYNTH_OPL_DBOPL_H
 #define SOUND_SOFTSYNTH_OPL_DBOPL_H
diff --git a/audio/softsynth/opl/dosbox.h b/audio/softsynth/opl/dosbox.h
index 125dde8aec..cdf86df114 100644
--- a/audio/softsynth/opl/dosbox.h
+++ b/audio/softsynth/opl/dosbox.h
@@ -104,4 +104,3 @@ public:
 #endif // !DISABLE_DOSBOX_OPL
 
 #endif
-
diff --git a/audio/softsynth/opl/mame.cpp b/audio/softsynth/opl/mame.cpp
index 74699ba4c6..dd3c354045 100644
--- a/audio/softsynth/opl/mame.cpp
+++ b/audio/softsynth/opl/mame.cpp
@@ -725,6 +725,8 @@ static int OPLOpenTable(void) {
 
 
 	ENV_CURVE = (int *)malloc(sizeof(int) * (2*EG_ENT+1));
+	if (!ENV_CURVE)
+		error("[OPLOpenTable] Cannot allocate memory");
 
 	/* envelope counter -> envelope output table */
 	for (i=0; i < EG_ENT; i++) {
@@ -1243,4 +1245,3 @@ FM_OPL *makeAdLibOPL(int rate) {
 
 } // End of namespace MAME
 } // End of namespace OPL
-