AUDIO: Rename sound/ dir to audio/

svn-id: r55850

22 files changed, 0 insertions, 4664 deletions

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