AUDIO: Split QuickTime audio into a new class

Standalone QuickTime files can now be played as an AudioStream

6 files changed, 4273 insertions, 0 deletions

diff --git a/audio/decoders/qdm2.cpp b/audio/decoders/qdm2.cpp
new file mode 100644
index 0000000000..2bbb608754
--- /dev/null
+++ b/audio/decoders/qdm2.cpp
@@ -0,0 +1,3281 @@
+/* ScummVM - Graphic Adventure Engine
+ *
+ * ScummVM is the legal property of its developers, whose names
+ * are too numerous to list here. Please refer to the COPYRIGHT
+ * file distributed with this source distribution.
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
+ *
+ * $URL$
+ * $Id$
+ *
+ */
+
+// Based off ffmpeg's QDM2 decoder
+
+#include "common/scummsys.h"
+#include "audio/decoders/qdm2.h"
+
+#ifdef AUDIO_QDM2_H
+
+#include "audio/audiostream.h"
+#include "audio/decoders/qdm2data.h"
+
+#include "common/array.h"
+#include "common/stream.h"
+#include "common/system.h"
+
+namespace Audio {
+
+enum {
+	SOFTCLIP_THRESHOLD = 27600,
+	HARDCLIP_THRESHOLD = 35716,
+	MPA_MAX_CHANNELS = 2,
+	MPA_FRAME_SIZE = 1152,
+	FF_INPUT_BUFFER_PADDING_SIZE = 8
+};
+
+typedef int8 sb_int8_array[2][30][64];
+
+/* bit input */
+/* buffer, buffer_end and size_in_bits must be present and used by every reader */
+struct GetBitContext {
+	const uint8 *buffer, *bufferEnd;
+	int index;
+	int sizeInBits;
+};
+
+struct QDM2SubPacket {
+	int type;
+	unsigned int size;
+	const uint8 *data; // pointer to subpacket data (points to input data buffer, it's not a private copy)
+};
+
+struct QDM2SubPNode {
+	QDM2SubPacket *packet;
+	struct QDM2SubPNode *next; // pointer to next packet in the list, NULL if leaf node
+};
+
+struct QDM2Complex {
+	float re;
+	float im;
+};
+
+struct FFTTone {
+	float level;
+	QDM2Complex *complex;
+	const float *table;
+	int phase;
+	int phase_shift;
+	int duration;
+	short time_index;
+	short cutoff;
+};
+
+struct FFTCoefficient {
+	int16 sub_packet;
+	uint8 channel;
+	int16 offset;
+	int16 exp;
+	uint8 phase;
+};
+
+struct VLC {
+	int32 bits;
+	int16 (*table)[2]; // code, bits
+	int32 table_size;
+	int32 table_allocated;
+};
+
+#include "common/pack-start.h"
+struct QDM2FFT {
+	QDM2Complex complex[MPA_MAX_CHANNELS][256];
+} PACKED_STRUCT;
+#include "common/pack-end.h"
+
+enum RDFTransformType {
+	RDFT,
+	IRDFT,
+	RIDFT,
+	IRIDFT
+};
+
+struct FFTComplex {
+	float re, im;
+};
+
+struct FFTContext {
+	int nbits;
+	int inverse;
+	uint16 *revtab;
+	FFTComplex *exptab;
+	FFTComplex *tmpBuf;
+	int mdctSize; // size of MDCT (i.e. number of input data * 2)
+	int mdctBits; // n = 2^nbits
+	// pre/post rotation tables
+	float *tcos;
+	float *tsin;
+	void (*fftPermute)(struct FFTContext *s, FFTComplex *z);
+	void (*fftCalc)(struct FFTContext *s, FFTComplex *z);
+	void (*imdctCalc)(struct FFTContext *s, float *output, const float *input);
+	void (*imdctHalf)(struct FFTContext *s, float *output, const float *input);
+	void (*mdctCalc)(struct FFTContext *s, float *output, const float *input);
+	int splitRadix;
+	int permutation;
+};
+
+enum {
+	FF_MDCT_PERM_NONE = 0,
+	FF_MDCT_PERM_INTERLEAVE = 1
+};
+
+struct RDFTContext {
+	int nbits;
+	int inverse;
+	int signConvention;
+
+	// pre/post rotation tables
+	float *tcos;
+	float *tsin;
+	FFTContext fft;
+};
+
+class QDM2Stream : public AudioStream {
+public:
+	QDM2Stream(Common::SeekableReadStream *stream, Common::SeekableReadStream *extraData);
+	~QDM2Stream();
+
+	bool isStereo() const { return _channels == 2; }
+	bool endOfData() const { return _stream->pos() >= _stream->size() && _outputSamples.size() == 0 && _subPacket == 0; }
+	int getRate() const { return _sampleRate; }
+	int readBuffer(int16 *buffer, const int numSamples);
+
+private:
+	Common::SeekableReadStream *_stream;
+
+	// Parameters from codec header, do not change during playback
+	uint8 _channels;
+	uint16 _sampleRate;
+	uint16 _bitRate;
+	uint16 _blockSize;  // Group
+	uint16 _frameSize;  // FFT
+	uint16 _packetSize; // Checksum
+
+	// Parameters built from header parameters, do not change during playback
+	int _groupOrder;       // order of frame group
+	int _fftOrder;         // order of FFT (actually fft order+1)
+	int _fftFrameSize;     // size of fft frame, in components (1 comples = re + im)
+	int _sFrameSize;        // size of data frame
+	int _frequencyRange;
+	int _subSampling;      // subsampling: 0=25%, 1=50%, 2=100% */
+	int _coeffPerSbSelect; // selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
+	int _cmTableSelect;    // selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
+
+	// Packets and packet lists
+	QDM2SubPacket _subPackets[16];    // the packets themselves
+	QDM2SubPNode _subPacketListA[16]; // list of all packets
+	QDM2SubPNode _subPacketListB[16]; // FFT packets B are on list
+	int _subPacketsB;                 // number of packets on 'B' list
+	QDM2SubPNode _subPacketListC[16]; // packets with errors?
+	QDM2SubPNode _subPacketListD[16]; // DCT packets
+
+	// FFT and tones
+	FFTTone _fftTones[1000];
+	int _fftToneStart;
+	int _fftToneEnd;
+	FFTCoefficient _fftCoefs[1000];
+	int _fftCoefsIndex;
+	int _fftCoefsMinIndex[5];
+	int _fftCoefsMaxIndex[5];
+	int _fftLevelExp[6];
+	RDFTContext _rdftCtx;
+	QDM2FFT _fft;
+
+	// I/O data
+	uint8 *_compressedData;
+	float _outputBuffer[1024];
+	Common::Array<int16> _outputSamples;
+
+	// Synthesis filter
+	int16 ff_mpa_synth_window[512];
+	int16 _synthBuf[MPA_MAX_CHANNELS][512*2];
+	int _synthBufOffset[MPA_MAX_CHANNELS];
+	int32 _sbSamples[MPA_MAX_CHANNELS][128][32];
+
+	// Mixed temporary data used in decoding
+	float _toneLevel[MPA_MAX_CHANNELS][30][64];
+	int8 _codingMethod[MPA_MAX_CHANNELS][30][64];
+	int8 _quantizedCoeffs[MPA_MAX_CHANNELS][10][8];
+	int8 _toneLevelIdxBase[MPA_MAX_CHANNELS][30][8];
+	int8 _toneLevelIdxHi1[MPA_MAX_CHANNELS][3][8][8];
+	int8 _toneLevelIdxMid[MPA_MAX_CHANNELS][26][8];
+	int8 _toneLevelIdxHi2[MPA_MAX_CHANNELS][26];
+	int8 _toneLevelIdx[MPA_MAX_CHANNELS][30][64];
+	int8 _toneLevelIdxTemp[MPA_MAX_CHANNELS][30][64];
+
+	// Flags
+	bool _hasErrors;         // packet has errors
+	int _superblocktype_2_3; // select fft tables and some algorithm based on superblock type
+	int _doSynthFilter;      // used to perform or skip synthesis filter
+
+	uint8 _subPacket; // 0 to 15
+	uint32 _superBlockStart;
+	int _noiseIdx; // index for dithering noise table
+
+	byte _emptyBuffer[FF_INPUT_BUFFER_PADDING_SIZE];
+
+	VLC _vlcTabLevel;
+	VLC _vlcTabDiff;
+	VLC _vlcTabRun;
+	VLC _fftLevelExpAltVlc;
+	VLC _fftLevelExpVlc;
+	VLC _fftStereoExpVlc;
+	VLC _fftStereoPhaseVlc;
+	VLC _vlcTabToneLevelIdxHi1;
+	VLC _vlcTabToneLevelIdxMid;
+	VLC _vlcTabToneLevelIdxHi2;
+	VLC _vlcTabType30;
+	VLC _vlcTabType34;
+	VLC _vlcTabFftToneOffset[5];
+	bool _vlcsInitialized;
+	void initVlc(void);
+
+	uint16 _softclipTable[HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1];
+	void softclipTableInit(void);
+
+	float _noiseTable[4096];
+	byte _randomDequantIndex[256][5];
+	byte _randomDequantType24[128][3];
+	void rndTableInit(void);
+
+	float _noiseSamples[128];
+	void initNoiseSamples(void);
+
+	void average_quantized_coeffs(void);
+	void build_sb_samples_from_noise(int sb);
+	void fix_coding_method_array(int sb, int channels, sb_int8_array coding_method);
+	void fill_tone_level_array(int flag);
+	void fill_coding_method_array(sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
+	                              sb_int8_array coding_method, int nb_channels,
+	                              int c, int superblocktype_2_3, int cm_table_select);
+	void synthfilt_build_sb_samples(GetBitContext *gb, int length, int sb_min, int sb_max);
+	void init_quantized_coeffs_elem0(int8 *quantized_coeffs, GetBitContext *gb, int length);
+	void init_tone_level_dequantization(GetBitContext *gb, int length);
+	void process_subpacket_9(QDM2SubPNode *node);
+	void process_subpacket_10(QDM2SubPNode *node, int length);
+	void process_subpacket_11(QDM2SubPNode *node, int length);
+	void process_subpacket_12(QDM2SubPNode *node, int length);
+	void process_synthesis_subpackets(QDM2SubPNode *list);
+	void qdm2_decode_super_block(void);
+	void qdm2_fft_init_coefficient(int sub_packet, int offset, int duration,
+	                               int channel, int exp, int phase);
+	void qdm2_fft_decode_tones(int duration, GetBitContext *gb, int b);
+	void qdm2_decode_fft_packets(void);
+	void qdm2_fft_generate_tone(FFTTone *tone);
+	void qdm2_fft_tone_synthesizer(uint8 sub_packet);
+	void qdm2_calculate_fft(int channel);
+	void qdm2_synthesis_filter(uint8 index);
+	int qdm2_decodeFrame(Common::SeekableReadStream *in);
+};
+
+// Fix compilation for non C99-compliant compilers, like MSVC
+#ifndef int64_t
+typedef signed long long int int64_t;
+#endif
+
+// Integer log2 function. This is much faster than invoking
+// double precision C99 log2 math functions or equivalent, since
+// this is only used to determine maximum number of bits needed
+// i.e. only non-fractional part is needed. Also, the double
+// version is incorrect for exact cases due to floating point
+// rounding errors.
+static inline int scummvm_log2(int n) {
+	int ret = -1;
+	while(n != 0) {
+		n /= 2;
+		ret++;
+	}
+	return ret;
+}
+
+#define QDM2_LIST_ADD(list, size, packet) \
+	do { \
+		if (size > 0) \
+			list[size - 1].next = &list[size]; \
+		list[size].packet = packet; \
+		list[size].next = NULL; \
+		size++; \
+	} while(0)
+
+// Result is 8, 16 or 30
+#define QDM2_SB_USED(subSampling) (((subSampling) >= 2) ? 30 : 8 << (subSampling))
+
+#define FIX_NOISE_IDX(noiseIdx) \
+	if ((noiseIdx) >= 3840) \
+		(noiseIdx) -= 3840 \
+
+#define SB_DITHERING_NOISE(sb, noiseIdx) (_noiseTable[(noiseIdx)++] * sb_noise_attenuation[(sb)])
+
+static inline void initGetBits(GetBitContext *s, const uint8 *buffer, int bitSize) {
+	int bufferSize = (bitSize + 7) >> 3;
+
+	if (bufferSize < 0 || bitSize < 0) {
+		bufferSize = bitSize = 0;
+		buffer = NULL;
+	}
+
+	s->buffer = buffer;
+	s->sizeInBits = bitSize;
+	s->bufferEnd = buffer + bufferSize;
+	s->index = 0;
+}
+
+static inline int getBitsCount(GetBitContext *s) {
+	return s->index;
+}
+
+static inline unsigned int getBits1(GetBitContext *s) {
+	int index;
+	uint8 result;
+
+	index = s->index;
+	result = s->buffer[index >> 3];
+
+	result >>= (index & 0x07);
+	result &= 1;
+	index++;
+	s->index = index;
+
+	return result;
+}
+
+static inline unsigned int getBits(GetBitContext *s, int n) {
+	int tmp, reCache, reIndex;
+
+	reIndex = s->index;
+
+	reCache = READ_LE_UINT32((const uint8 *)s->buffer + (reIndex >> 3)) >> (reIndex & 0x07);
+
+	tmp = (reCache) & ((uint32)0xffffffff >> (32 - n));
+
+	s->index = reIndex + n;
+
+	return tmp;
+}
+
+static inline void skipBits(GetBitContext *s, int n) {
+	s->index += n;
+}
+
+#define BITS_LEFT(length, gb) ((length) - getBitsCount((gb)))
+
+static int splitRadixPermutation(int i, int n, int inverse) {
+	if (n <= 2)
+		return i & 1;
+
+	int m = n >> 1;
+
+	if(!(i & m))
+		return splitRadixPermutation(i, m, inverse) * 2;
+
+	m >>= 1;
+
+	if (inverse == !(i & m))
+		return splitRadixPermutation(i, m, inverse) * 4 + 1;
+
+	return splitRadixPermutation(i, m, inverse) * 4 - 1;
+}
+
+// sin(2*pi*x/n) for 0<=x<n/4, followed by n/2<=x<3n/4
+float ff_sin_16[8];
+float ff_sin_32[16];
+float ff_sin_64[32];
+float ff_sin_128[64];
+float ff_sin_256[128];
+float ff_sin_512[256];
+float ff_sin_1024[512];
+float ff_sin_2048[1024];
+float ff_sin_4096[2048];
+float ff_sin_8192[4096];
+float ff_sin_16384[8192];
+float ff_sin_32768[16384];
+float ff_sin_65536[32768];
+
+float *ff_sin_tabs[] = {
+	NULL, NULL, NULL, NULL,
+	ff_sin_16, ff_sin_32, ff_sin_64, ff_sin_128, ff_sin_256, ff_sin_512, ff_sin_1024,
+	ff_sin_2048, ff_sin_4096, ff_sin_8192, ff_sin_16384, ff_sin_32768, ff_sin_65536,
+};
+
+// cos(2*pi*x/n) for 0<=x<=n/4, followed by its reverse
+float ff_cos_16[8];
+float ff_cos_32[16];
+float ff_cos_64[32];
+float ff_cos_128[64];
+float ff_cos_256[128];
+float ff_cos_512[256];
+float ff_cos_1024[512];
+float ff_cos_2048[1024];
+float ff_cos_4096[2048];
+float ff_cos_8192[4096];
+float ff_cos_16384[8192];
+float ff_cos_32768[16384];
+float ff_cos_65536[32768];
+
+float *ff_cos_tabs[] = {
+	NULL, NULL, NULL, NULL,
+	ff_cos_16, ff_cos_32, ff_cos_64, ff_cos_128, ff_cos_256, ff_cos_512, ff_cos_1024,
+	ff_cos_2048, ff_cos_4096, ff_cos_8192, ff_cos_16384, ff_cos_32768, ff_cos_65536,
+};
+
+void initCosineTables(int index) {
+	int m = 1 << index;
+	double freq = 2 * M_PI / m;
+	float *tab = ff_cos_tabs[index];
+
+	for (int i = 0; i <= m / 4; i++)
+		tab[i] = cos(i * freq);
+
+	for (int i = 1; i < m / 4; i++)
+		tab[m / 2 - i] = tab[i];
+}
+
+void fftPermute(FFTContext *s, FFTComplex *z) {
+	const uint16 *revtab = s->revtab;
+	int np = 1 << s->nbits;
+
+	if (s->tmpBuf) {
+		// TODO: handle split-radix permute in a more optimal way, probably in-place
+		for (int j = 0; j < np; j++)
+			s->tmpBuf[revtab[j]] = z[j];
+		memcpy(z, s->tmpBuf, np * sizeof(FFTComplex));
+		return;
+	}
+
+	// reverse
+	for (int j = 0; j < np; j++) {
+		int k = revtab[j];
+		if (k < j) {
+			FFTComplex tmp = z[k];
+			z[k] = z[j];
+			z[j] = tmp;
+		}
+	}
+}
+
+#define DECL_FFT(n,n2,n4) \
+static void fft##n(FFTComplex *z) { \
+	fft##n2(z); \
+	fft##n4(z + n4 * 2); \
+	fft##n4(z + n4 * 3); \
+	pass(z, ff_cos_##n, n4 / 2); \
+}
+
+#ifndef M_SQRT1_2
+#define M_SQRT1_2 7.0710678118654752440E-1
+#endif
+
+#define sqrthalf (float)M_SQRT1_2
+
+#define BF(x,y,a,b) { \
+	x = a - b; \
+	y = a + b; \
+}
+
+#define BUTTERFLIES(a0, a1, a2, a3) { \
+	BF(t3, t5, t5, t1); \
+	BF(a2.re, a0.re, a0.re, t5); \
+	BF(a3.im, a1.im, a1.im, t3); \
+	BF(t4, t6, t2, t6); \
+	BF(a3.re, a1.re, a1.re, t4); \
+	BF(a2.im, a0.im, a0.im, t6); \
+}
+
+// force loading all the inputs before storing any.
+// this is slightly slower for small data, but avoids store->load aliasing
+// for addresses separated by large powers of 2.
+#define BUTTERFLIES_BIG(a0, a1, a2, a3) { \
+	float r0 = a0.re, i0 = a0.im, r1 = a1.re, i1 = a1.im; \
+	BF(t3, t5, t5, t1); \
+	BF(a2.re, a0.re, r0, t5); \
+	BF(a3.im, a1.im, i1, t3); \
+	BF(t4, t6, t2, t6); \
+	BF(a3.re, a1.re, r1, t4); \
+	BF(a2.im, a0.im, i0, t6); \
+}
+
+#define TRANSFORM(a0, a1, a2, a3, wre, wim) { \
+	t1 = a2.re * wre + a2.im * wim; \
+	t2 = a2.im * wre - a2.re * wim; \
+	t5 = a3.re * wre - a3.im * wim; \
+	t6 = a3.im * wre + a3.re * wim; \
+	BUTTERFLIES(a0, a1, a2, a3) \
+}
+
+#define TRANSFORM_ZERO(a0, a1, a2, a3) { \
+	t1 = a2.re; \
+	t2 = a2.im; \
+	t5 = a3.re; \
+	t6 = a3.im; \
+	BUTTERFLIES(a0, a1, a2, a3) \
+}
+
+// z[0...8n-1], w[1...2n-1]
+#define PASS(name) \
+static void name(FFTComplex *z, const float *wre, unsigned int n) { \
+	float t1, t2, t3, t4, t5, t6; \
+	int o1 = 2 * n; \
+	int o2 = 4 * n; \
+	int o3 = 6 * n; \
+	const float *wim = wre + o1; \
+	n--; \
+	\
+	TRANSFORM_ZERO(z[0], z[o1], z[o2], z[o3]); \
+	TRANSFORM(z[1], z[o1 + 1], z[o2 + 1], z[o3 + 1], wre[1], wim[-1]); \
+	\
+	do { \
+		z += 2; \
+		wre += 2; \
+		wim -= 2; \
+		TRANSFORM(z[0], z[o1], z[o2], z[o3], wre[0], wim[0]); \
+		TRANSFORM(z[1], z[o1 + 1],z[o2 + 1], z[o3 + 1], wre[1], wim[-1]); \
+	} while(--n); \
+}
+
+PASS(pass)
+#undef BUTTERFLIES
+#define BUTTERFLIES BUTTERFLIES_BIG
+PASS(pass_big)
+
+static void fft4(FFTComplex *z) {
+	float t1, t2, t3, t4, t5, t6, t7, t8;
+
+	BF(t3, t1, z[0].re, z[1].re);
+	BF(t8, t6, z[3].re, z[2].re);
+	BF(z[2].re, z[0].re, t1, t6);
+	BF(t4, t2, z[0].im, z[1].im);
+	BF(t7, t5, z[2].im, z[3].im);
+	BF(z[3].im, z[1].im, t4, t8);
+	BF(z[3].re, z[1].re, t3, t7);
+	BF(z[2].im, z[0].im, t2, t5);
+}
+
+static void fft8(FFTComplex *z) {
+	float t1, t2, t3, t4, t5, t6, t7, t8;
+
+	fft4(z);
+
+	BF(t1, z[5].re, z[4].re, -z[5].re);
+	BF(t2, z[5].im, z[4].im, -z[5].im);
+	BF(t3, z[7].re, z[6].re, -z[7].re);
+	BF(t4, z[7].im, z[6].im, -z[7].im);
+	BF(t8, t1, t3, t1);
+	BF(t7, t2, t2, t4);
+	BF(z[4].re, z[0].re, z[0].re, t1);
+	BF(z[4].im, z[0].im, z[0].im, t2);
+	BF(z[6].re, z[2].re, z[2].re, t7);
+	BF(z[6].im, z[2].im, z[2].im, t8);
+
+	TRANSFORM(z[1], z[3], z[5], z[7], sqrthalf, sqrthalf);
+}
+
+#undef BF
+
+DECL_FFT(16,8,4)
+DECL_FFT(32,16,8)
+DECL_FFT(64,32,16)
+DECL_FFT(128,64,32)
+DECL_FFT(256,128,64)
+DECL_FFT(512,256,128)
+#define pass pass_big
+DECL_FFT(1024,512,256)
+DECL_FFT(2048,1024,512)
+DECL_FFT(4096,2048,1024)
+DECL_FFT(8192,4096,2048)
+DECL_FFT(16384,8192,4096)
+DECL_FFT(32768,16384,8192)
+DECL_FFT(65536,32768,16384)
+
+void fftCalc(FFTContext *s, FFTComplex *z) {
+	static void (* const fftDispatch[])(FFTComplex*) = {
+		fft4, fft8, fft16, fft32, fft64, fft128, fft256, fft512, fft1024,
+		fft2048, fft4096, fft8192, fft16384, fft32768, fft65536,
+	};
+
+	fftDispatch[s->nbits - 2](z);
+}
+
+// complex multiplication: p = a * b
+#define CMUL(pre, pim, are, aim, bre, bim) \
+{\
+	float _are = (are); \
+	float _aim = (aim); \
+	float _bre = (bre); \
+	float _bim = (bim); \
+	(pre) = _are * _bre - _aim * _bim; \
+	(pim) = _are * _bim + _aim * _bre; \
+}
+
+/**
+ * Compute the middle half of the inverse MDCT of size N = 2^nbits,
+ * thus excluding the parts that can be derived by symmetry
+ * @param output N/2 samples
+ * @param input N/2 samples
+ */
+void imdctHalfC(FFTContext *s, float *output, const float *input) {
+	const uint16 *revtab = s->revtab;
+	const float *tcos = s->tcos;
+	const float *tsin = s->tsin;
+	FFTComplex *z = (FFTComplex *)output;
+
+	int n = 1 << s->mdctBits;
+	int n2 = n >> 1;
+	int n4 = n >> 2;
+	int n8 = n >> 3;
+
+	// pre rotation
+	const float *in1 = input;
+	const float *in2 = input + n2 - 1;
+	for (int k = 0; k < n4; k++) {
+		int j = revtab[k];
+		CMUL(z[j].re, z[j].im, *in2, *in1, tcos[k], tsin[k]);
+		in1 += 2;
+		in2 -= 2;
+	}
+
+	fftCalc(s, z);
+
+	// post rotation + reordering
+	for (int k = 0; k < n8; k++) {
+		float r0, i0, r1, i1;
+		CMUL(r0, i1, z[n8 - k - 1].im, z[n8 - k - 1].re, tsin[n8 - k - 1], tcos[n8 - k - 1]);
+		CMUL(r1, i0, z[n8 + k].im, z[n8 + k].re, tsin[n8 + k], tcos[n8 + k]);
+		z[n8 - k - 1].re = r0;
+		z[n8 - k - 1].im = i0;
+		z[n8 + k].re = r1;
+		z[n8 + k].im = i1;
+	}
+}
+
+/**
+ * Compute inverse MDCT of size N = 2^nbits
+ * @param output N samples
+ * @param input N/2 samples
+ */
+void imdctCalcC(FFTContext *s, float *output, const float *input) {
+	int n = 1 << s->mdctBits;
+	int n2 = n >> 1;
+	int n4 = n >> 2;
+
+	imdctHalfC(s, output + n4, input);
+
+	for (int k = 0; k < n4; k++) {
+		output[k] = -output[n2 - k - 1];
+		output[n - k - 1] = output[n2 + k];
+	}
+}
+
+/**
+ * Compute MDCT of size N = 2^nbits
+ * @param input N samples
+ * @param out N/2 samples
+ */
+void mdctCalcC(FFTContext *s, float *out, const float *input) {
+	const uint16 *revtab = s->revtab;
+	const float *tcos = s->tcos;
+	const float *tsin = s->tsin;
+	FFTComplex *x = (FFTComplex *)out;
+
+	int n = 1 << s->mdctBits;
+	int n2 = n >> 1;
+	int n4 = n >> 2;
+	int n8 = n >> 3;
+	int n3 = 3 * n4;
+
+	// pre rotation
+	for (int i = 0; i < n8; i++) {
+		float re = -input[2 * i + 3 * n4] - input[n3 - 1 - 2 * i];
+		float im = -input[n4 + 2 * i] + input[n4 - 1 - 2 * i];
+		int j = revtab[i];
+		CMUL(x[j].re, x[j].im, re, im, -tcos[i], tsin[i]);
+
+		re = input[2 * i] - input[n2 - 1 - 2 * i];
+		im = -(input[n2 + 2 * i] + input[n - 1 - 2 * i]);
+		j = revtab[n8 + i];
+		CMUL(x[j].re, x[j].im, re, im, -tcos[n8 + i], tsin[n8 + i]);
+	}
+
+	fftCalc(s, x);
+
+	// post rotation
+	for (int i = 0; i < n8; i++) {
+		float r0, i0, r1, i1;
+		CMUL(i1, r0, x[n8 - i - 1].re, x[n8 - i - 1].im, -tsin[n8 - i - 1], -tcos[n8 - i - 1]);
+		CMUL(i0, r1, x[n8 + i].re, x[n8 + i].im, -tsin[n8 + i], -tcos[n8 + i]);
+		x[n8 - i - 1].re = r0;
+		x[n8 - i - 1].im = i0;
+		x[n8 + i].re = r1;
+		x[n8 + i].im = i1;
+	}
+}
+
+int fftInit(FFTContext *s, int nbits, int inverse) {
+	int i, j, m, n;
+	float alpha, c1, s1, s2;
+
+	if (nbits < 2 || nbits > 16)
+		goto fail;
+
+	s->nbits = nbits;
+	n = 1 << nbits;
+	s->tmpBuf = NULL;
+
+	s->exptab = (FFTComplex *)malloc((n / 2) * sizeof(FFTComplex));
+	if (!s->exptab)
+		goto fail;
+
+	s->revtab = (uint16 *)malloc(n * sizeof(uint16));
+	if (!s->revtab)
+		goto fail;
+	s->inverse = inverse;
+
+	s2 = inverse ? 1.0 : -1.0;
+
+	s->fftPermute = fftPermute;
+	s->fftCalc = fftCalc;
+	s->imdctCalc = imdctCalcC;
+	s->imdctHalf = imdctHalfC;
+	s->mdctCalc = mdctCalcC;
+	s->splitRadix = 1;
+
+	if (s->splitRadix) {
+		for (j = 4; j <= nbits; j++)
+			initCosineTables(j);
+
+		for (i = 0; i < n; i++)
+			s->revtab[-splitRadixPermutation(i, n, s->inverse) & (n - 1)] = i;
+
+		s->tmpBuf = (FFTComplex *)malloc(n * sizeof(FFTComplex));
+	} else {
+		for (i = 0; i < n / 2; i++) {
+			alpha = 2 * M_PI * (float)i / (float)n;
+			c1 = cos(alpha);
+			s1 = sin(alpha) * s2;
+			s->exptab[i].re = c1;
+			s->exptab[i].im = s1;
+		}
+
+		//int np = 1 << nbits;
+		//int nblocks = np >> 3;
+		//int np2 = np >> 1;
+
+		// compute bit reverse table
+		for (i = 0; i < n; i++) {
+			m = 0;
+
+			for (j = 0; j < nbits; j++)
+				m |= ((i >> j) & 1) << (nbits - j - 1);
+
+			s->revtab[i] = m;
+		}
+	}
+
+	return 0;
+
+ fail:
+	free(&s->revtab);
+	free(&s->exptab);
+	free(&s->tmpBuf);
+	return -1;
+}
+
+/**
+ * Sets up a real FFT.
+ * @param nbits           log2 of the length of the input array
+ * @param trans           the type of transform
+ */
+int rdftInit(RDFTContext *s, int nbits, RDFTransformType trans) {
+	int n = 1 << nbits;
+	const double theta = (trans == RDFT || trans == IRIDFT ? -1 : 1) * 2 * M_PI / n;
+
+	s->nbits = nbits;
+	s->inverse = trans == IRDFT || trans == IRIDFT;
+	s->signConvention = trans == RIDFT || trans == IRIDFT ? 1 : -1;
+
+	if (nbits < 4 || nbits > 16)
+		return -1;
+
+	if (fftInit(&s->fft, nbits - 1, trans == IRDFT || trans == RIDFT) < 0)
+		return -1;
+
+	initCosineTables(nbits);
+	s->tcos = ff_cos_tabs[nbits];
+	s->tsin = ff_sin_tabs[nbits] + (trans == RDFT || trans == IRIDFT) * (n >> 2);
+
+	for (int i = 0; i < n >> 2; i++)
+		s->tsin[i] = sin(i*theta);
+
+	return 0;
+}
+
+/** Map one real FFT into two parallel real even and odd FFTs. Then interleave
+ * the two real FFTs into one complex FFT. Unmangle the results.
+ * ref: http://www.engineeringproductivitytools.com/stuff/T0001/PT10.HTM
+ */
+void rdftCalc(RDFTContext *s, float *data) {
+	FFTComplex ev, od;
+
+	const int n = 1 << s->nbits;
+	const float k1 = 0.5;
+	const float k2 = 0.5 - s->inverse;
+	const float *tcos = s->tcos;
+	const float *tsin = s->tsin;
+
+	if (!s->inverse) {
+		fftPermute(&s->fft, (FFTComplex *)data);
+		fftCalc(&s->fft, (FFTComplex *)data);
+	}
+
+	// i=0 is a special case because of packing, the DC term is real, so we
+	// are going to throw the N/2 term (also real) in with it.
+	ev.re = data[0];
+	data[0] = ev.re + data[1];
+	data[1] = ev.re - data[1];
+
+	int i;
+
+	for (i = 1; i < n >> 2; i++) {
+		int i1 = i * 2;
+		int i2 = n - i1;
+
+		// Separate even and odd FFTs
+		ev.re = k1 * (data[i1] + data[i2]);
+		od.im = -k2 * (data[i1] - data[i2]);
+		ev.im = k1 * (data[i1 + 1] - data[i2 + 1]);
+		od.re = k2 * (data[i1 + 1] + data[i2 + 1]);
+
+		// Apply twiddle factors to the odd FFT and add to the even FFT
+		data[i1] = ev.re + od.re * tcos[i] - od.im * tsin[i];
+		data[i1 + 1] = ev.im + od.im * tcos[i] + od.re * tsin[i];
+		data[i2] = ev.re - od.re * tcos[i] + od.im * tsin[i];
+		data[i2 + 1] = -ev.im + od.im * tcos[i] + od.re * tsin[i];
+	}
+
+	data[i * 2 + 1] = s->signConvention * data[i * 2 + 1];
+	if (s->inverse) {
+		data[0] *= k1;
+		data[1] *= k1;
+		fftPermute(&s->fft, (FFTComplex*)data);
+		fftCalc(&s->fft, (FFTComplex*)data);
+	}
+}
+
+// half mpeg encoding window (full precision)
+const int32 ff_mpa_enwindow[257] = {
+     0,    -1,    -1,    -1,    -1,    -1,    -1,    -2,
+    -2,    -2,    -2,    -3,    -3,    -4,    -4,    -5,
+    -5,    -6,    -7,    -7,    -8,    -9,   -10,   -11,
+   -13,   -14,   -16,   -17,   -19,   -21,   -24,   -26,
+   -29,   -31,   -35,   -38,   -41,   -45,   -49,   -53,
+   -58,   -63,   -68,   -73,   -79,   -85,   -91,   -97,
+  -104,  -111,  -117,  -125,  -132,  -139,  -147,  -154,
+  -161,  -169,  -176,  -183,  -190,  -196,  -202,  -208,
+   213,   218,   222,   225,   227,   228,   228,   227,
+   224,   221,   215,   208,   200,   189,   177,   163,
+   146,   127,   106,    83,    57,    29,    -2,   -36,
+   -72,  -111,  -153,  -197,  -244,  -294,  -347,  -401,
+  -459,  -519,  -581,  -645,  -711,  -779,  -848,  -919,
+  -991, -1064, -1137, -1210, -1283, -1356, -1428, -1498,
+ -1567, -1634, -1698, -1759, -1817, -1870, -1919, -1962,
+ -2001, -2032, -2057, -2075, -2085, -2087, -2080, -2063,
+  2037,  2000,  1952,  1893,  1822,  1739,  1644,  1535,
+  1414,  1280,  1131,   970,   794,   605,   402,   185,
+   -45,  -288,  -545,  -814, -1095, -1388, -1692, -2006,
+ -2330, -2663, -3004, -3351, -3705, -4063, -4425, -4788,
+ -5153, -5517, -5879, -6237, -6589, -6935, -7271, -7597,
+ -7910, -8209, -8491, -8755, -8998, -9219, -9416, -9585,
+ -9727, -9838, -9916, -9959, -9966, -9935, -9863, -9750,
+ -9592, -9389, -9139, -8840, -8492, -8092, -7640, -7134,
+  6574,  5959,  5288,  4561,  3776,  2935,  2037,  1082,
+    70,  -998, -2122, -3300, -4533, -5818, -7154, -8540,
+ -9975,-11455,-12980,-14548,-16155,-17799,-19478,-21189,
+-22929,-24694,-26482,-28289,-30112,-31947,-33791,-35640,
+-37489,-39336,-41176,-43006,-44821,-46617,-48390,-50137,
+-51853,-53534,-55178,-56778,-58333,-59838,-61289,-62684,
+-64019,-65290,-66494,-67629,-68692,-69679,-70590,-71420,
+-72169,-72835,-73415,-73908,-74313,-74630,-74856,-74992,
+ 75038
+};
+
+void ff_mpa_synth_init(int16 *window) {
+	int i;
+	int32 v;
+
+	// max = 18760, max sum over all 16 coefs : 44736
+	for(i = 0; i < 257; i++) {
+		v = ff_mpa_enwindow[i];
+		v = (v + 2) >> 2;
+		window[i] = v;
+
+		if ((i & 63) != 0)
+			v = -v;
+
+		if (i != 0)
+			window[512 - i] = v;
+	}
+}
+
+static inline uint16 round_sample(int *sum) {
+	int sum1;
+	sum1 = (*sum) >> 14;
+	*sum &= (1 << 14)-1;
+	if (sum1 < (-0x7fff - 1))
+		sum1 = (-0x7fff - 1);
+	if (sum1 > 0x7fff)
+		sum1 = 0x7fff;
+	return sum1;
+}
+
+static inline int MULH(int a, int b) {
+	return ((int64_t)(a) * (int64_t)(b))>>32;
+}
+
+// signed 16x16 -> 32 multiply add accumulate
+#define MACS(rt, ra, rb) rt += (ra) * (rb)
+
+#define MLSS(rt, ra, rb) ((rt) -= (ra) * (rb))
+
+#define SUM8(op, sum, w, p)\
+{\
+	op(sum, (w)[0 * 64], (p)[0 * 64]);\
+	op(sum, (w)[1 * 64], (p)[1 * 64]);\
+	op(sum, (w)[2 * 64], (p)[2 * 64]);\
+	op(sum, (w)[3 * 64], (p)[3 * 64]);\
+	op(sum, (w)[4 * 64], (p)[4 * 64]);\
+	op(sum, (w)[5 * 64], (p)[5 * 64]);\
+	op(sum, (w)[6 * 64], (p)[6 * 64]);\
+	op(sum, (w)[7 * 64], (p)[7 * 64]);\
+}
+
+#define SUM8P2(sum1, op1, sum2, op2, w1, w2, p) \
+{\
+	tmp_s = p[0 * 64];\
+	op1(sum1, (w1)[0 * 64], tmp_s);\
+	op2(sum2, (w2)[0 * 64], tmp_s);\
+	tmp_s = p[1 * 64];\
+	op1(sum1, (w1)[1 * 64], tmp_s);\
+	op2(sum2, (w2)[1 * 64], tmp_s);\
+	tmp_s = p[2 * 64];\
+	op1(sum1, (w1)[2 * 64], tmp_s);\
+	op2(sum2, (w2)[2 * 64], tmp_s);\
+	tmp_s = p[3 * 64];\
+	op1(sum1, (w1)[3 * 64], tmp_s);\
+	op2(sum2, (w2)[3 * 64], tmp_s);\
+	tmp_s = p[4 * 64];\
+	op1(sum1, (w1)[4 * 64], tmp_s);\
+	op2(sum2, (w2)[4 * 64], tmp_s);\
+	tmp_s = p[5 * 64];\
+	op1(sum1, (w1)[5 * 64], tmp_s);\
+	op2(sum2, (w2)[5 * 64], tmp_s);\
+	tmp_s = p[6 * 64];\
+	op1(sum1, (w1)[6 * 64], tmp_s);\
+	op2(sum2, (w2)[6 * 64], tmp_s);\
+	tmp_s = p[7 * 64];\
+	op1(sum1, (w1)[7 * 64], tmp_s);\
+	op2(sum2, (w2)[7 * 64], tmp_s);\
+}
+
+#define FIXHR(a) ((int)((a) * (1LL<<32) + 0.5))
+
+// tab[i][j] = 1.0 / (2.0 * cos(pi*(2*k+1) / 2^(6 - j)))
+
+// cos(i*pi/64)
+
+#define COS0_0  FIXHR(0.50060299823519630134/2)
+#define COS0_1  FIXHR(0.50547095989754365998/2)
+#define COS0_2  FIXHR(0.51544730992262454697/2)
+#define COS0_3  FIXHR(0.53104259108978417447/2)
+#define COS0_4  FIXHR(0.55310389603444452782/2)
+#define COS0_5  FIXHR(0.58293496820613387367/2)
+#define COS0_6  FIXHR(0.62250412303566481615/2)
+#define COS0_7  FIXHR(0.67480834145500574602/2)
+#define COS0_8  FIXHR(0.74453627100229844977/2)
+#define COS0_9  FIXHR(0.83934964541552703873/2)
+#define COS0_10 FIXHR(0.97256823786196069369/2)
+#define COS0_11 FIXHR(1.16943993343288495515/4)
+#define COS0_12 FIXHR(1.48416461631416627724/4)
+#define COS0_13 FIXHR(2.05778100995341155085/8)
+#define COS0_14 FIXHR(3.40760841846871878570/8)
+#define COS0_15 FIXHR(10.19000812354805681150/32)
+
+#define COS1_0 FIXHR(0.50241928618815570551/2)
+#define COS1_1 FIXHR(0.52249861493968888062/2)
+#define COS1_2 FIXHR(0.56694403481635770368/2)
+#define COS1_3 FIXHR(0.64682178335999012954/2)
+#define COS1_4 FIXHR(0.78815462345125022473/2)
+#define COS1_5 FIXHR(1.06067768599034747134/4)
+#define COS1_6 FIXHR(1.72244709823833392782/4)
+#define COS1_7 FIXHR(5.10114861868916385802/16)
+
+#define COS2_0 FIXHR(0.50979557910415916894/2)
+#define COS2_1 FIXHR(0.60134488693504528054/2)
+#define COS2_2 FIXHR(0.89997622313641570463/2)
+#define COS2_3 FIXHR(2.56291544774150617881/8)
+
+#define COS3_0 FIXHR(0.54119610014619698439/2)
+#define COS3_1 FIXHR(1.30656296487637652785/4)
+
+#define COS4_0 FIXHR(0.70710678118654752439/2)
+
+/* butterfly operator */
+#define BF(a, b, c, s)\
+{\
+	tmp0 = tab[a] + tab[b];\
+	tmp1 = tab[a] - tab[b];\
+	tab[a] = tmp0;\
+	tab[b] = MULH(tmp1<<(s), c);\
+}
+
+#define BF1(a, b, c, d)\
+{\
+	BF(a, b, COS4_0, 1);\
+	BF(c, d,-COS4_0, 1);\
+	tab[c] += tab[d];\
+}
+
+#define BF2(a, b, c, d)\
+{\
+	BF(a, b, COS4_0, 1);\
+	BF(c, d,-COS4_0, 1);\
+	tab[c] += tab[d];\
+	tab[a] += tab[c];\
+	tab[c] += tab[b];\
+	tab[b] += tab[d];\
+}
+
+#define ADD(a, b) tab[a] += tab[b]
+
+// DCT32 without 1/sqrt(2) coef zero scaling.
+static void dct32(int32 *out, int32 *tab) {
+	int tmp0, tmp1;
+
+	// pass 1
+	BF( 0, 31, COS0_0 , 1);
+	BF(15, 16, COS0_15, 5);
+	// pass 2
+	BF( 0, 15, COS1_0 , 1);
+	BF(16, 31,-COS1_0 , 1);
+	// pass 1
+	BF( 7, 24, COS0_7 , 1);
+	BF( 8, 23, COS0_8 , 1);
+	// pass 2
+	BF( 7,  8, COS1_7 , 4);
+	BF(23, 24,-COS1_7 , 4);
+	// pass 3
+	BF( 0,  7, COS2_0 , 1);
+	BF( 8, 15,-COS2_0 , 1);
+	BF(16, 23, COS2_0 , 1);
+	BF(24, 31,-COS2_0 , 1);
+	// pass 1
+	BF( 3, 28, COS0_3 , 1);
+	BF(12, 19, COS0_12, 2);
+	// pass 2
+	BF( 3, 12, COS1_3 , 1);
+	BF(19, 28,-COS1_3 , 1);
+	// pass 1
+	BF( 4, 27, COS0_4 , 1);
+	BF(11, 20, COS0_11, 2);
+	// pass 2
+	BF( 4, 11, COS1_4 , 1);
+	BF(20, 27,-COS1_4 , 1);
+	// pass 3
+	BF( 3,  4, COS2_3 , 3);
+	BF(11, 12,-COS2_3 , 3);
+	BF(19, 20, COS2_3 , 3);
+	BF(27, 28,-COS2_3 , 3);
+	// pass 4
+	BF( 0,  3, COS3_0 , 1);
+	BF( 4,  7,-COS3_0 , 1);
+	BF( 8, 11, COS3_0 , 1);
+	BF(12, 15,-COS3_0 , 1);
+	BF(16, 19, COS3_0 , 1);
+	BF(20, 23,-COS3_0 , 1);
+	BF(24, 27, COS3_0 , 1);
+	BF(28, 31,-COS3_0 , 1);
+
+	// pass 1
+	BF( 1, 30, COS0_1 , 1);
+	BF(14, 17, COS0_14, 3);
+	// pass 2
+	BF( 1, 14, COS1_1 , 1);
+	BF(17, 30,-COS1_1 , 1);
+	// pass 1
+	BF( 6, 25, COS0_6 , 1);
+	BF( 9, 22, COS0_9 , 1);
+	// pass 2
+	BF( 6,  9, COS1_6 , 2);
+	BF(22, 25,-COS1_6 , 2);
+	// pass 3
+	BF( 1,  6, COS2_1 , 1);
+	BF( 9, 14,-COS2_1 , 1);
+	BF(17, 22, COS2_1 , 1);
+	BF(25, 30,-COS2_1 , 1);
+
+	// pass 1
+	BF( 2, 29, COS0_2 , 1);
+	BF(13, 18, COS0_13, 3);
+	// pass 2
+	BF( 2, 13, COS1_2 , 1);
+	BF(18, 29,-COS1_2 , 1);
+	// pass 1
+	BF( 5, 26, COS0_5 , 1);
+	BF(10, 21, COS0_10, 1);
+	// pass 2
+	BF( 5, 10, COS1_5 , 2);
+	BF(21, 26,-COS1_5 , 2);
+	// pass 3
+	BF( 2,  5, COS2_2 , 1);
+	BF(10, 13,-COS2_2 , 1);
+	BF(18, 21, COS2_2 , 1);
+	BF(26, 29,-COS2_2 , 1);
+	// pass 4
+	BF( 1,  2, COS3_1 , 2);
+	BF( 5,  6,-COS3_1 , 2);
+	BF( 9, 10, COS3_1 , 2);
+	BF(13, 14,-COS3_1 , 2);
+	BF(17, 18, COS3_1 , 2);
+	BF(21, 22,-COS3_1 , 2);
+	BF(25, 26, COS3_1 , 2);
+	BF(29, 30,-COS3_1 , 2);
+
+	// pass 5
+	BF1( 0,  1,  2,  3);
+	BF2( 4,  5,  6,  7);
+	BF1( 8,  9, 10, 11);
+	BF2(12, 13, 14, 15);
+	BF1(16, 17, 18, 19);
+	BF2(20, 21, 22, 23);
+	BF1(24, 25, 26, 27);
+	BF2(28, 29, 30, 31);
+
+	// pass 6
+	ADD( 8, 12);
+	ADD(12, 10);
+	ADD(10, 14);
+	ADD(14,  9);
+	ADD( 9, 13);
+	ADD(13, 11);
+	ADD(11, 15);
+
+	out[ 0] = tab[0];
+	out[16] = tab[1];
+	out[ 8] = tab[2];
+	out[24] = tab[3];
+	out[ 4] = tab[4];
+	out[20] = tab[5];
+	out[12] = tab[6];
+	out[28] = tab[7];
+	out[ 2] = tab[8];
+	out[18] = tab[9];
+	out[10] = tab[10];
+	out[26] = tab[11];
+	out[ 6] = tab[12];
+	out[22] = tab[13];
+	out[14] = tab[14];
+	out[30] = tab[15];
+
+	ADD(24, 28);
+	ADD(28, 26);
+	ADD(26, 30);
+	ADD(30, 25);
+	ADD(25, 29);
+	ADD(29, 27);
+	ADD(27, 31);
+
+	out[ 1] = tab[16] + tab[24];
+	out[17] = tab[17] + tab[25];
+	out[ 9] = tab[18] + tab[26];
+	out[25] = tab[19] + tab[27];
+	out[ 5] = tab[20] + tab[28];
+	out[21] = tab[21] + tab[29];
+	out[13] = tab[22] + tab[30];
+	out[29] = tab[23] + tab[31];
+	out[ 3] = tab[24] + tab[20];
+	out[19] = tab[25] + tab[21];
+	out[11] = tab[26] + tab[22];
+	out[27] = tab[27] + tab[23];
+	out[ 7] = tab[28] + tab[18];
+	out[23] = tab[29] + tab[19];
+	out[15] = tab[30] + tab[17];
+	out[31] = tab[31];
+}
+
+// 32 sub band synthesis filter. Input: 32 sub band samples, Output:
+// 32 samples.
+// XXX: optimize by avoiding ring buffer usage
+void ff_mpa_synth_filter(int16 *synth_buf_ptr, int *synth_buf_offset,
+                         int16 *window, int *dither_state,
+                         int16 *samples, int incr,
+                         int32 sb_samples[32])
+{
+	int16 *synth_buf;
+	const int16 *w, *w2, *p;
+	int j, offset;
+	int16 *samples2;
+	int32 tmp[32];
+	int sum, sum2;
+	int tmp_s;
+
+	offset = *synth_buf_offset;
+	synth_buf = synth_buf_ptr + offset;
+
+	dct32(tmp, sb_samples);
+	for(j = 0; j < 32; j++) {
+		// NOTE: can cause a loss in precision if very high amplitude sound
+		if (tmp[j] < (-0x7fff - 1))
+			synth_buf[j] = (-0x7fff - 1);
+		else if (tmp[j] > 0x7fff)
+			synth_buf[j] = 0x7fff;
+		else
+			synth_buf[j] = tmp[j];
+	}
+
+	// copy to avoid wrap
+	memcpy(synth_buf + 512, synth_buf, 32 * sizeof(int16));
+
+	samples2 = samples + 31 * incr;
+	w = window;
+	w2 = window + 31;
+
+	sum = *dither_state;
+	p = synth_buf + 16;
+	SUM8(MACS, sum, w, p);
+	p = synth_buf + 48;
+	SUM8(MLSS, sum, w + 32, p);
+	*samples = round_sample(&sum);
+	samples += incr;
+	w++;
+
+	// we calculate two samples at the same time to avoid one memory
+	// access per two sample
+	for(j = 1; j < 16; j++) {
+		sum2 = 0;
+		p = synth_buf + 16 + j;
+		SUM8P2(sum, MACS, sum2, MLSS, w, w2, p);
+		p = synth_buf + 48 - j;
+		SUM8P2(sum, MLSS, sum2, MLSS, w + 32, w2 + 32, p);
+
+		*samples = round_sample(&sum);
+		samples += incr;
+		sum += sum2;
+		*samples2 = round_sample(&sum);
+		samples2 -= incr;
+		w++;
+		w2--;
+	}
+
+	p = synth_buf + 32;
+	SUM8(MLSS, sum, w + 32, p);
+	*samples = round_sample(&sum);
+	*dither_state= sum;
+
+	offset = (offset - 32) & 511;
+	*synth_buf_offset = offset;
+}
+
+/**
+ * parses a vlc code, faster then get_vlc()
+ * @param bits is the number of bits which will be read at once, must be
+ *             identical to nb_bits in init_vlc()
+ * @param max_depth is the number of times bits bits must be read to completely
+ *                  read the longest vlc code
+ *                  = (max_vlc_length + bits - 1) / bits
+ */
+static int getVlc2(GetBitContext *s, int16 (*table)[2], int bits, int maxDepth) {
+	int reIndex;
+	int reCache;
+	int index;
+	int code;
+	int n;
+
+	reIndex = s->index;
+	reCache = READ_LE_UINT32(s->buffer + (reIndex >> 3)) >> (reIndex & 0x07);
+	index = reCache & (0xffffffff >> (32 - bits));
+	code = table[index][0];
+	n = table[index][1];
+
+	if (maxDepth > 1 && n < 0){
+		reIndex += bits;
+		reCache = READ_LE_UINT32(s->buffer + (reIndex >> 3)) >> (reIndex & 0x07);
+
+		int nbBits = -n;
+
+		index = (reCache & (0xffffffff >> (32 - nbBits))) + code;
+		code = table[index][0];
+		n = table[index][1];
+
+		if(maxDepth > 2 && n < 0) {
+			reIndex += nbBits;
+			reCache = READ_LE_UINT32(s->buffer + (reIndex >> 3)) >> (reIndex & 0x07);
+
+			nbBits = -n;
+
+			index = (reCache & (0xffffffff >> (32 - nbBits))) + code;
+			code = table[index][0];
+			n = table[index][1];
+		}
+	}
+
+	reCache >>= n;
+	s->index = reIndex + n;
+	return code;
+}
+
+static int allocTable(VLC *vlc, int size, int use_static) {
+	int index;
+	int16 (*temp)[2] = NULL;
+	index = vlc->table_size;
+	vlc->table_size += size;
+	if (vlc->table_size > vlc->table_allocated) {
+		if(use_static)
+			error("QDM2 cant do anything, init_vlc() is used with too little memory");
+		vlc->table_allocated += (1 << vlc->bits);
+		temp = (int16 (*)[2])realloc(vlc->table, sizeof(int16 *) * 2 * vlc->table_allocated);
+		if (!temp) {
+			free(vlc->table);
+			vlc->table = NULL;
+			return -1;
+		}
+		vlc->table = temp;
+	}
+	return index;
+}
+
+#define GET_DATA(v, table, i, wrap, size)\
+{\
+	const uint8 *ptr = (const uint8 *)table + i * wrap;\
+	switch(size) {\
+		case 1:\
+			v = *(const uint8 *)ptr;\
+			break;\
+		case 2:\
+			v = *(const uint16 *)ptr;\
+			break;\
+		default:\
+			v = *(const uint32 *)ptr;\
+			break;\
+	}\
+}
+
+static int build_table(VLC *vlc, int table_nb_bits,
+                       int nb_codes,
+                       const void *bits, int bits_wrap, int bits_size,
+                       const void *codes, int codes_wrap, int codes_size,
+                       const void *symbols, int symbols_wrap, int symbols_size,
+                       int code_prefix, int n_prefix, int flags)
+{
+	int i, j, k, n, table_size, table_index, nb, n1, index, code_prefix2, symbol;
+	uint32 code;
+	int16 (*table)[2];
+
+	table_size = 1 << table_nb_bits;
+	table_index = allocTable(vlc, table_size, flags & 4);
+	if (table_index < 0)
+		return -1;
+	table = &vlc->table[table_index];
+
+	for(i = 0; i < table_size; i++) {
+		table[i][1] = 0; //bits
+		table[i][0] = -1; //codes
+	}
+
+	// first pass: map codes and compute auxillary table sizes
+	for(i = 0; i < nb_codes; i++) {
+		GET_DATA(n, bits, i, bits_wrap, bits_size);
+		GET_DATA(code, codes, i, codes_wrap, codes_size);
+		// we accept tables with holes
+		if (n <= 0)
+			continue;
+		if (!symbols)
+			symbol = i;
+		else
+			GET_DATA(symbol, symbols, i, symbols_wrap, symbols_size);
+		// if code matches the prefix, it is in the table
+		n -= n_prefix;
+		if(flags & 2)
+			code_prefix2= code & (n_prefix>=32 ? 0xffffffff : (1 << n_prefix)-1);
+		else
+			code_prefix2= code >> n;
+		if (n > 0 && code_prefix2 == code_prefix) {
+			if (n <= table_nb_bits) {
+				// no need to add another table
+				j = (code << (table_nb_bits - n)) & (table_size - 1);
+				nb = 1 << (table_nb_bits - n);
+				for(k = 0; k < nb; k++) {
+					if(flags & 2)
+						j = (code >> n_prefix) + (k<<n);
+					if (table[j][1] /*bits*/ != 0) {
+						error("QDM2 incorrect codes");
+						return -1;
+					}
+					table[j][1] = n; //bits
+					table[j][0] = symbol;
+					j++;
+				}
+			} else {
+				n -= table_nb_bits;
+				j = (code >> ((flags & 2) ? n_prefix : n)) & ((1 << table_nb_bits) - 1);
+				// compute table size
+				n1 = -table[j][1]; //bits
+				if (n > n1)
+					n1 = n;
+				table[j][1] = -n1; //bits
+			}
+		}
+	}
+
+	// second pass : fill auxillary tables recursively
+	for(i = 0;i < table_size; i++) {
+		n = table[i][1]; //bits
+		if (n < 0) {
+			n = -n;
+			if (n > table_nb_bits) {
+				n = table_nb_bits;
+				table[i][1] = -n; //bits
+			}
+			index = build_table(vlc, n, nb_codes,
+			                    bits, bits_wrap, bits_size,
+			                    codes, codes_wrap, codes_size,
+			                    symbols, symbols_wrap, symbols_size,
+			                    (flags & 2) ? (code_prefix | (i << n_prefix)) : ((code_prefix << table_nb_bits) | i),
+			                    n_prefix + table_nb_bits, flags);
+ 			if (index < 0)
+				return -1;
+			// note: realloc has been done, so reload tables
+			table = &vlc->table[table_index];
+			table[i][0] = index; //code
+		}
+	}
+	return table_index;
+}
+
+/* Build VLC decoding tables suitable for use with get_vlc().
+
+   'nb_bits' set thee decoding table size (2^nb_bits) entries. The
+   bigger it is, the faster is the decoding. But it should not be too
+   big to save memory and L1 cache. '9' is a good compromise.
+
+   'nb_codes' : number of vlcs codes
+
+   'bits' : table which gives the size (in bits) of each vlc code.
+
+   'codes' : table which gives the bit pattern of of each vlc code.
+
+   'symbols' : table which gives the values to be returned from get_vlc().
+
+   'xxx_wrap' : give the number of bytes between each entry of the
+   'bits' or 'codes' tables.
+
+   'xxx_size' : gives the number of bytes of each entry of the 'bits'
+   or 'codes' tables.
+
+   'wrap' and 'size' allows to use any memory configuration and types
+   (byte/word/long) to store the 'bits', 'codes', and 'symbols' tables.
+
+   'use_static' should be set to 1 for tables, which should be freed
+   with av_free_static(), 0 if free_vlc() will be used.
+*/
+void initVlcSparse(VLC *vlc, int nb_bits, int nb_codes,
+		const void *bits, int bits_wrap, int bits_size,
+		const void *codes, int codes_wrap, int codes_size,
+		const void *symbols, int symbols_wrap, int symbols_size) {
+	vlc->bits = nb_bits;
+
+	if(vlc->table_size && vlc->table_size == vlc->table_allocated) {
+		return;
+	} else if(vlc->table_size) {
+		error("called on a partially initialized table");
+	}
+
+	if (build_table(vlc, nb_bits, nb_codes,
+	                bits, bits_wrap, bits_size,
+	                codes, codes_wrap, codes_size,
+	                symbols, symbols_wrap, symbols_size,
+	                0, 0, 4 | 2) < 0) {
+		free(&vlc->table);
+		return; // Error
+	}
+
+	if(vlc->table_size != vlc->table_allocated)
+		error("QDM2 needed %d had %d", vlc->table_size, vlc->table_allocated);
+}
+
+void QDM2Stream::softclipTableInit(void) {
+	uint16 i;
+	double dfl = SOFTCLIP_THRESHOLD - 32767;
+	float delta = 1.0 / -dfl;
+
+	for (i = 0; i < ARRAYSIZE(_softclipTable); i++)
+		_softclipTable[i] = SOFTCLIP_THRESHOLD - ((int)(sin((float)i * delta) * dfl) & 0x0000FFFF);
+}
+
+// random generated table
+void QDM2Stream::rndTableInit(void) {
+	uint16 i;
+	uint16 j;
+	uint32 ldw, hdw;
+	// TODO: Replace Code with uint64 less version...
+	int64_t tmp64_1;
+	int64_t random_seed = 0;
+	float delta = 1.0 / 16384.0;
+
+	for(i = 0; i < ARRAYSIZE(_noiseTable); i++) {
+		random_seed = random_seed * 214013 + 2531011;
+		_noiseTable[i] = (delta * (float)(((int32)random_seed >> 16) & 0x00007FFF)- 1.0) * 1.3;
+	}
+
+	for (i = 0; i < 256; i++) {
+		random_seed = 81;
+		ldw = i;
+		for (j = 0; j < 5; j++) {
+			_randomDequantIndex[i][j] = (uint8)((ldw / random_seed) & 0xFF);
+			ldw = (uint32)ldw % (uint32)random_seed;
+			tmp64_1 = (random_seed * 0x55555556);
+			hdw = (uint32)(tmp64_1 >> 32);
+			random_seed = (int64_t)(hdw + (ldw >> 31));
+		}
+	}
+
+	for (i = 0; i < 128; i++) {
+		random_seed = 25;
+		ldw = i;
+		for (j = 0; j < 3; j++) {
+			_randomDequantType24[i][j] = (uint8)((ldw / random_seed) & 0xFF);
+			ldw = (uint32)ldw % (uint32)random_seed;
+			tmp64_1 = (random_seed * 0x66666667);
+			hdw = (uint32)(tmp64_1 >> 33);
+			random_seed = hdw + (ldw >> 31);
+		}
+	}
+}
+
+void QDM2Stream::initNoiseSamples(void) {
+	uint16 i;
+	uint32 random_seed = 0;
+	float delta = 1.0 / 16384.0;
+
+	for (i = 0; i < ARRAYSIZE(_noiseSamples); i++) {
+		random_seed = random_seed * 214013 + 2531011;
+		_noiseSamples[i] = (delta * (float)((random_seed >> 16) & 0x00007fff) - 1.0);
+	}
+}
+
+static const uint16 qdm2_vlc_offs[18] = {
+	0, 260, 566, 598, 894, 1166, 1230, 1294, 1678, 1950, 2214, 2278, 2310, 2570, 2834, 3124, 3448, 3838
+};
+
+void QDM2Stream::initVlc(void) {
+	static int16 qdm2_table[3838][2];
+
+	if (!_vlcsInitialized) {
+		_vlcTabLevel.table = &qdm2_table[qdm2_vlc_offs[0]];
+		_vlcTabLevel.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0];
+		_vlcTabLevel.table_size = 0;
+		initVlcSparse(&_vlcTabLevel, 8, 24,
+			vlc_tab_level_huffbits, 1, 1,
+			vlc_tab_level_huffcodes, 2, 2, NULL, 0, 0);
+
+		_vlcTabDiff.table = &qdm2_table[qdm2_vlc_offs[1]];
+		_vlcTabDiff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1];
+		_vlcTabDiff.table_size = 0;
+		initVlcSparse(&_vlcTabDiff, 8, 37,
+			vlc_tab_diff_huffbits, 1, 1,
+			vlc_tab_diff_huffcodes, 2, 2, NULL, 0, 0);
+
+		_vlcTabRun.table = &qdm2_table[qdm2_vlc_offs[2]];
+		_vlcTabRun.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2];
+		_vlcTabRun.table_size = 0;
+		initVlcSparse(&_vlcTabRun, 5, 6,
+			vlc_tab_run_huffbits, 1, 1,
+			vlc_tab_run_huffcodes, 1, 1, NULL, 0, 0);
+
+		_fftLevelExpAltVlc.table = &qdm2_table[qdm2_vlc_offs[3]];
+		_fftLevelExpAltVlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3];
+		_fftLevelExpAltVlc.table_size = 0;
+		initVlcSparse(&_fftLevelExpAltVlc, 8, 28,
+			fft_level_exp_alt_huffbits, 1, 1,
+			fft_level_exp_alt_huffcodes, 2, 2, NULL, 0, 0);
+
+		_fftLevelExpVlc.table = &qdm2_table[qdm2_vlc_offs[4]];
+		_fftLevelExpVlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4];
+		_fftLevelExpVlc.table_size = 0;
+		initVlcSparse(&_fftLevelExpVlc, 8, 20,
+			fft_level_exp_huffbits, 1, 1,
+			fft_level_exp_huffcodes, 2, 2, NULL, 0, 0);
+
+		_fftStereoExpVlc.table = &qdm2_table[qdm2_vlc_offs[5]];
+		_fftStereoExpVlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5];
+		_fftStereoExpVlc.table_size = 0;
+		initVlcSparse(&_fftStereoExpVlc, 6, 7,
+			fft_stereo_exp_huffbits, 1, 1,
+			fft_stereo_exp_huffcodes, 1, 1, NULL, 0, 0);
+
+		_fftStereoPhaseVlc.table = &qdm2_table[qdm2_vlc_offs[6]];
+		_fftStereoPhaseVlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6];
+		_fftStereoPhaseVlc.table_size = 0;
+		initVlcSparse(&_fftStereoPhaseVlc, 6, 9,
+			fft_stereo_phase_huffbits, 1, 1,
+			fft_stereo_phase_huffcodes, 1, 1, NULL, 0, 0);
+
+		_vlcTabToneLevelIdxHi1.table = &qdm2_table[qdm2_vlc_offs[7]];
+		_vlcTabToneLevelIdxHi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7];
+		_vlcTabToneLevelIdxHi1.table_size = 0;
+		initVlcSparse(&_vlcTabToneLevelIdxHi1, 8, 20,
+			vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
+			vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, NULL, 0, 0);
+
+		_vlcTabToneLevelIdxMid.table = &qdm2_table[qdm2_vlc_offs[8]];
+		_vlcTabToneLevelIdxMid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8];
+		_vlcTabToneLevelIdxMid.table_size = 0;
+		initVlcSparse(&_vlcTabToneLevelIdxMid, 8, 24,
+			vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
+			vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, NULL, 0, 0);
+
+		_vlcTabToneLevelIdxHi2.table = &qdm2_table[qdm2_vlc_offs[9]];
+		_vlcTabToneLevelIdxHi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9];
+		_vlcTabToneLevelIdxHi2.table_size = 0;
+		initVlcSparse(&_vlcTabToneLevelIdxHi2, 8, 24,
+			vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
+			vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, NULL, 0, 0);
+
+		_vlcTabType30.table = &qdm2_table[qdm2_vlc_offs[10]];
+		_vlcTabType30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10];
+		_vlcTabType30.table_size = 0;
+		initVlcSparse(&_vlcTabType30, 6, 9,
+			vlc_tab_type30_huffbits, 1, 1,
+			vlc_tab_type30_huffcodes, 1, 1, NULL, 0, 0);
+
+		_vlcTabType34.table = &qdm2_table[qdm2_vlc_offs[11]];
+		_vlcTabType34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11];
+		_vlcTabType34.table_size = 0;
+		initVlcSparse(&_vlcTabType34, 5, 10,
+			vlc_tab_type34_huffbits, 1, 1,
+			vlc_tab_type34_huffcodes, 1, 1, NULL, 0, 0);
+
+		_vlcTabFftToneOffset[0].table = &qdm2_table[qdm2_vlc_offs[12]];
+		_vlcTabFftToneOffset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12];
+		_vlcTabFftToneOffset[0].table_size = 0;
+		initVlcSparse(&_vlcTabFftToneOffset[0], 8, 23,
+			vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
+			vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, NULL, 0, 0);
+
+		_vlcTabFftToneOffset[1].table = &qdm2_table[qdm2_vlc_offs[13]];
+		_vlcTabFftToneOffset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13];
+		_vlcTabFftToneOffset[1].table_size = 0;
+		initVlcSparse(&_vlcTabFftToneOffset[1], 8, 28,
+			vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
+			vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, NULL, 0, 0);
+
+		_vlcTabFftToneOffset[2].table = &qdm2_table[qdm2_vlc_offs[14]];
+		_vlcTabFftToneOffset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14];
+		_vlcTabFftToneOffset[2].table_size = 0;
+		initVlcSparse(&_vlcTabFftToneOffset[2], 8, 32,
+			vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
+			vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, NULL, 0, 0);
+
+		_vlcTabFftToneOffset[3].table = &qdm2_table[qdm2_vlc_offs[15]];
+		_vlcTabFftToneOffset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15];
+		_vlcTabFftToneOffset[3].table_size = 0;
+		initVlcSparse(&_vlcTabFftToneOffset[3], 8, 35,
+			vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
+			vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, NULL, 0, 0);
+
+		_vlcTabFftToneOffset[4].table = &qdm2_table[qdm2_vlc_offs[16]];
+		_vlcTabFftToneOffset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16];
+		_vlcTabFftToneOffset[4].table_size = 0;
+		initVlcSparse(&_vlcTabFftToneOffset[4], 8, 38,
+			vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
+			vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, NULL, 0, 0);
+
+		_vlcsInitialized = true;
+	}
+}
+
+QDM2Stream::QDM2Stream(Common::SeekableReadStream *stream, Common::SeekableReadStream *extraData) {
+	uint32 tmp;
+	int32 tmp_s;
+	int tmp_val;
+	int i;
+
+	debug(1, "QDM2Stream::QDM2Stream() Call");
+
+	_stream = stream;
+	_compressedData = NULL;
+	_subPacket = 0;
+	_superBlockStart = 0;
+	memset(_quantizedCoeffs, 0, sizeof(_quantizedCoeffs));
+	memset(_fftLevelExp, 0, sizeof(_fftLevelExp));
+	_noiseIdx = 0;
+	memset(_fftCoefsMinIndex, 0, sizeof(_fftCoefsMinIndex));
+	memset(_fftCoefsMaxIndex, 0, sizeof(_fftCoefsMaxIndex));
+	_fftToneStart = 0;
+	_fftToneEnd = 0;
+	for(i = 0; i < ARRAYSIZE(_subPacketListA); i++) {
+		_subPacketListA[i].packet = NULL;
+		_subPacketListA[i].next = NULL;
+	}
+	_subPacketsB = 0;
+	for(i = 0; i < ARRAYSIZE(_subPacketListB); i++) {
+		_subPacketListB[i].packet = NULL;
+		_subPacketListB[i].next = NULL;
+	}
+	for(i = 0; i < ARRAYSIZE(_subPacketListC); i++) {
+		_subPacketListC[i].packet = NULL;
+		_subPacketListC[i].next = NULL;
+	}
+	for(i = 0; i < ARRAYSIZE(_subPacketListD); i++) {
+		_subPacketListD[i].packet = NULL;
+		_subPacketListD[i].next = NULL;
+	}
+	memset(_synthBuf, 0, sizeof(_synthBuf));
+	memset(_synthBufOffset, 0, sizeof(_synthBufOffset));
+	memset(_sbSamples, 0, sizeof(_sbSamples));
+	memset(_outputBuffer, 0, sizeof(_outputBuffer));
+	_vlcsInitialized = false;
+	_superblocktype_2_3 = 0;
+	_hasErrors = false;
+
+	// Rewind extraData stream from any previous calls...
+	extraData->seek(0, SEEK_SET);
+
+	tmp_s = extraData->readSint32BE();
+	debug(1, "QDM2Stream::QDM2Stream() extraSize: %d", tmp_s);
+	if ((extraData->size() - extraData->pos()) / 4 + 1 != tmp_s)
+		warning("QDM2Stream::QDM2Stream() extraSize mismatch - Expected %d", (extraData->size() - extraData->pos()) / 4 + 1);
+	if (tmp_s < 12)
+		error("QDM2Stream::QDM2Stream() Insufficient extraData");
+
+	tmp = extraData->readUint32BE();
+	debug(1, "QDM2Stream::QDM2Stream() extraTag: %d", tmp);
+	if (tmp != MKID_BE('frma'))
+		warning("QDM2Stream::QDM2Stream() extraTag mismatch");
+
+	tmp = extraData->readUint32BE();
+	debug(1, "QDM2Stream::QDM2Stream() extraType: %d", tmp);
+	if (tmp == MKID_BE('QDMC'))
+		warning("QDM2Stream::QDM2Stream() QDMC stream type not supported");
+	else if (tmp != MKID_BE('QDM2'))
+		error("QDM2Stream::QDM2Stream() Unsupported stream type");
+
+	tmp_s = extraData->readSint32BE();
+	debug(1, "QDM2Stream::QDM2Stream() extraSize2: %d", tmp_s);
+	if ((extraData->size() - extraData->pos()) + 4 != tmp_s)
+		warning("QDM2Stream::QDM2Stream() extraSize2 mismatch - Expected %d", (extraData->size() - extraData->pos()) + 4);
+
+	tmp = extraData->readUint32BE();
+	debug(1, "QDM2Stream::QDM2Stream() extraTag2: %d", tmp);
+	if (tmp != MKID_BE('QDCA'))
+		warning("QDM2Stream::QDM2Stream() extraTag2 mismatch");
+
+	if (extraData->readUint32BE() != 1)
+		warning("QDM2Stream::QDM2Stream() u0 field not 1");
+
+	_channels = extraData->readUint32BE();
+	debug(1, "QDM2Stream::QDM2Stream() channels: %d", _channels);
+
+	_sampleRate = extraData->readUint32BE();
+	debug(1, "QDM2Stream::QDM2Stream() sampleRate: %d", _sampleRate);
+
+	_bitRate = extraData->readUint32BE();
+	debug(1, "QDM2Stream::QDM2Stream() bitRate: %d", _bitRate);
+
+	_blockSize = extraData->readUint32BE();
+	debug(1, "QDM2Stream::QDM2Stream() blockSize: %d", _blockSize);
+
+	_frameSize = extraData->readUint32BE();
+	debug(1, "QDM2Stream::QDM2Stream() frameSize: %d", _frameSize);
+
+	_packetSize = extraData->readUint32BE();
+	debug(1, "QDM2Stream::QDM2Stream() packetSize: %d", _packetSize);
+
+	if (extraData->size() - extraData->pos() != 0) {
+		tmp_s = extraData->readSint32BE();
+		debug(1, "QDM2Stream::QDM2Stream() extraSize3: %d", tmp_s);
+		if (extraData->size() + 4 != tmp_s)
+			warning("QDM2Stream::QDM2Stream() extraSize3 mismatch - Expected %d", extraData->size() + 4);
+
+		tmp = extraData->readUint32BE();
+		debug(1, "QDM2Stream::QDM2Stream() extraTag3: %d", tmp);
+		if (tmp != MKID_BE('QDCP'))
+			warning("QDM2Stream::QDM2Stream() extraTag3 mismatch");
+
+		if ((float)extraData->readUint32BE() != 1.0)
+			warning("QDM2Stream::QDM2Stream() uf0 field not 1.0");
+
+		if (extraData->readUint32BE() != 0)
+			warning("QDM2Stream::QDM2Stream() u1 field not 0");
+
+		if ((float)extraData->readUint32BE() != 1.0)
+			warning("QDM2Stream::QDM2Stream() uf1 field not 1.0");
+
+		if ((float)extraData->readUint32BE() != 1.0)
+			warning("QDM2Stream::QDM2Stream() uf2 field not 1.0");
+
+		if (extraData->readUint32BE() != 27)
+			warning("QDM2Stream::QDM2Stream() u2 field not 27");
+
+		if (extraData->readUint32BE() != 8)
+			warning("QDM2Stream::QDM2Stream() u3 field not 8");
+
+		if (extraData->readUint32BE() != 0)
+			warning("QDM2Stream::QDM2Stream() u4 field not 0");
+	}
+
+	_fftOrder = scummvm_log2(_frameSize) + 1;
+	_fftFrameSize = 2 * _frameSize; // complex has two floats
+
+	// something like max decodable tones
+	_groupOrder = scummvm_log2(_blockSize) + 1;
+	_sFrameSize = _blockSize / 16; // 16 iterations per super block
+
+	_subSampling = _fftOrder - 7;
+	_frequencyRange = 255 / (1 << (2 - _subSampling));
+
+	switch ((_subSampling * 2 + _channels - 1)) {
+		case 0:
+			tmp = 40;
+			break;
+		case 1:
+			tmp = 48;
+			break;
+		case 2:
+			tmp = 56;
+			break;
+		case 3:
+			tmp = 72;
+			break;
+		case 4:
+			tmp = 80;
+			break;
+		case 5:
+			tmp = 100;
+			break;
+		default:
+			tmp = _subSampling;
+			break;
+	}
+
+	tmp_val = 0;
+	if ((tmp * 1000) < _bitRate)  tmp_val = 1;
+	if ((tmp * 1440) < _bitRate)  tmp_val = 2;
+	if ((tmp * 1760) < _bitRate)  tmp_val = 3;
+	if ((tmp * 2240) < _bitRate)  tmp_val = 4;
+	_cmTableSelect = tmp_val;
+
+	if (_subSampling == 0)
+		tmp = 7999;
+	else
+		tmp = ((-(_subSampling -1)) & 8000) + 20000;
+
+	if (tmp < 8000)
+		_coeffPerSbSelect = 0;
+	else if (tmp <= 16000)
+		_coeffPerSbSelect = 1;
+	else
+		_coeffPerSbSelect = 2;
+
+	if (_fftOrder < 7 || _fftOrder > 9)
+		error("QDM2Stream::QDM2Stream() Unsupported fft_order: %d", _fftOrder);
+
+	rdftInit(&_rdftCtx, _fftOrder, IRDFT);
+
+	initVlc();
+	ff_mpa_synth_init(ff_mpa_synth_window);
+	softclipTableInit();
+	rndTableInit();
+	initNoiseSamples();
+
+	_compressedData = new uint8[_packetSize];
+}
+
+QDM2Stream::~QDM2Stream() {
+	delete[] _compressedData;
+	delete _stream;
+}
+
+static int qdm2_get_vlc(GetBitContext *gb, VLC *vlc, int flag, int depth) {
+	int value = getVlc2(gb, vlc->table, vlc->bits, depth);
+
+	// stage-2, 3 bits exponent escape sequence
+	if (value-- == 0)
+		value = getBits(gb, getBits (gb, 3) + 1);
+
+	// stage-3, optional
+	if (flag) {
+		int tmp = vlc_stage3_values[value];
+
+		if ((value & ~3) > 0)
+			tmp += getBits(gb, (value >> 2));
+		value = tmp;
+	}
+
+	return value;
+}
+
+static int qdm2_get_se_vlc(VLC *vlc, GetBitContext *gb, int depth)
+{
+	int value = qdm2_get_vlc(gb, vlc, 0, depth);
+
+	return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
+}
+
+/**
+ * QDM2 checksum
+ *
+ * @param data      pointer to data to be checksum'ed
+ * @param length    data length
+ * @param value     checksum value
+ *
+ * @return          0 if checksum is OK
+ */
+static uint16 qdm2_packet_checksum(const uint8 *data, int length, int value) {
+	int i;
+
+	for (i = 0; i < length; i++)
+		value -= data[i];
+
+	return (uint16)(value & 0xffff);
+}
+
+/**
+ * Fills a QDM2SubPacket structure with packet type, size, and data pointer.
+ *
+ * @param gb            bitreader context
+ * @param sub_packet    packet under analysis
+ */
+static void qdm2_decode_sub_packet_header(GetBitContext *gb, QDM2SubPacket *sub_packet)
+{
+	sub_packet->type = getBits (gb, 8);
+
+	if (sub_packet->type == 0) {
+		sub_packet->size = 0;
+		sub_packet->data = NULL;
+	} else {
+		sub_packet->size = getBits (gb, 8);
+
+		if (sub_packet->type & 0x80) {
+			sub_packet->size <<= 8;
+			sub_packet->size  |= getBits (gb, 8);
+			sub_packet->type  &= 0x7f;
+		}
+
+		if (sub_packet->type == 0x7f)
+			sub_packet->type |= (getBits (gb, 8) << 8);
+
+		sub_packet->data = &gb->buffer[getBitsCount(gb) / 8]; // FIXME: this depends on bitreader internal data
+	}
+}
+
+/**
+ * Return node pointer to first packet of requested type in list.
+ *
+ * @param list    list of subpackets to be scanned
+ * @param type    type of searched subpacket
+ * @return        node pointer for subpacket if found, else NULL
+ */
+static QDM2SubPNode* qdm2_search_subpacket_type_in_list(QDM2SubPNode *list, int type)
+{
+	while (list != NULL && list->packet != NULL) {
+		if (list->packet->type == type)
+			return list;
+		list = list->next;
+	}
+	return NULL;
+}
+
+/**
+ * Replaces 8 elements with their average value.
+ * Called by qdm2_decode_superblock before starting subblock decoding.
+ */
+void QDM2Stream::average_quantized_coeffs(void) {
+	int i, j, n, ch, sum;
+
+	n = coeff_per_sb_for_avg[_coeffPerSbSelect][QDM2_SB_USED(_subSampling) - 1] + 1;
+
+	for (ch = 0; ch < _channels; ch++) {
+		for (i = 0; i < n; i++) {
+			sum = 0;
+
+			for (j = 0; j < 8; j++)
+				sum += _quantizedCoeffs[ch][i][j];
+
+			sum /= 8;
+			if (sum > 0)
+				sum--;
+
+			for (j = 0; j < 8; j++)
+				_quantizedCoeffs[ch][i][j] = sum;
+		}
+	}
+}
+
+/**
+ * Build subband samples with noise weighted by q->tone_level.
+ * Called by synthfilt_build_sb_samples.
+ *
+ * @param sb    subband index
+ */
+void QDM2Stream::build_sb_samples_from_noise(int sb) {
+	int ch, j;
+
+	FIX_NOISE_IDX(_noiseIdx);
+
+	if (!_channels)
+		return;
+
+	for (ch = 0; ch < _channels; ch++) {
+		for (j = 0; j < 64; j++) {
+			_sbSamples[ch][j * 2][sb] = (int32)(SB_DITHERING_NOISE(sb, _noiseIdx) * _toneLevel[ch][sb][j] + .5);
+			_sbSamples[ch][j * 2 + 1][sb] = (int32)(SB_DITHERING_NOISE(sb, _noiseIdx) * _toneLevel[ch][sb][j] + .5);
+		}
+	}
+}
+
+/**
+ * Called while processing data from subpackets 11 and 12.
+ * Used after making changes to coding_method array.
+ *
+ * @param sb               subband index
+ * @param channels         number of channels
+ * @param coding_method    q->coding_method[0][0][0]
+ */
+void QDM2Stream::fix_coding_method_array(int sb, int channels, sb_int8_array coding_method)
+{
+	int j, k;
+	int ch;
+	int run, case_val;
+	int switchtable[23] = {0,5,1,5,5,5,5,5,2,5,5,5,5,5,5,5,3,5,5,5,5,5,4};
+
+	for (ch = 0; ch < channels; ch++) {
+		for (j = 0; j < 64; ) {
+			if((coding_method[ch][sb][j] - 8) > 22) {
+				run = 1;
+				case_val = 8;
+			} else {
+				switch (switchtable[coding_method[ch][sb][j]-8]) {
+					case 0: run = 10; case_val = 10; break;
+					case 1: run = 1; case_val = 16; break;
+					case 2: run = 5; case_val = 24; break;
+					case 3: run = 3; case_val = 30; break;
+					case 4: run = 1; case_val = 30; break;
+					case 5: run = 1; case_val = 8; break;
+					default: run = 1; case_val = 8; break;
+				}
+			}
+			for (k = 0; k < run; k++)
+				if (j + k < 128)
+					if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
+						if (k > 0) {
+							warning("QDM2 Untested Code: not debugged, almost never used");
+							memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8));
+							memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8));
+						}
+			j += run;
+		}
+	}
+}
+
+/**
+ * Related to synthesis filter
+ * Called by process_subpacket_10
+ *
+ * @param flag    1 if called after getting data from subpacket 10, 0 if no subpacket 10
+ */
+void QDM2Stream::fill_tone_level_array(int flag) {
+	int i, sb, ch, sb_used;
+	int tmp, tab;
+
+	// This should never happen
+	if (_channels <= 0)
+		return;
+
+	for (ch = 0; ch < _channels; ch++) {
+		for (sb = 0; sb < 30; sb++) {
+			for (i = 0; i < 8; i++) {
+				if ((tab=coeff_per_sb_for_dequant[_coeffPerSbSelect][sb]) < (last_coeff[_coeffPerSbSelect] - 1))
+					tmp = _quantizedCoeffs[ch][tab + 1][i] * dequant_table[_coeffPerSbSelect][tab + 1][sb]+
+					      _quantizedCoeffs[ch][tab][i] * dequant_table[_coeffPerSbSelect][tab][sb];
+				else
+					tmp = _quantizedCoeffs[ch][tab][i] * dequant_table[_coeffPerSbSelect][tab][sb];
+				if(tmp < 0)
+					tmp += 0xff;
+				_toneLevelIdxBase[ch][sb][i] = (tmp / 256) & 0xff;
+			}
+		}
+	}
+
+	sb_used = QDM2_SB_USED(_subSampling);
+
+	if ((_superblocktype_2_3 != 0) && !flag) {
+		for (sb = 0; sb < sb_used; sb++) {
+			for (ch = 0; ch < _channels; ch++) {
+				for (i = 0; i < 64; i++) {
+					_toneLevelIdx[ch][sb][i] = _toneLevelIdxBase[ch][sb][i / 8];
+					if (_toneLevelIdx[ch][sb][i] < 0)
+						_toneLevel[ch][sb][i] = 0;
+					else
+						_toneLevel[ch][sb][i] = fft_tone_level_table[0][_toneLevelIdx[ch][sb][i] & 0x3f];
+				}
+			}
+		}
+	} else {
+		tab = _superblocktype_2_3 ? 0 : 1;
+		for (sb = 0; sb < sb_used; sb++) {
+			if ((sb >= 4) && (sb <= 23)) {
+				for (ch = 0; ch < _channels; ch++) {
+					for (i = 0; i < 64; i++) {
+						tmp = _toneLevelIdxBase[ch][sb][i / 8] -
+						      _toneLevelIdxHi1[ch][sb / 8][i / 8][i % 8] -
+						      _toneLevelIdxMid[ch][sb - 4][i / 8] -
+						      _toneLevelIdxHi2[ch][sb - 4];
+						_toneLevelIdx[ch][sb][i] = tmp & 0xff;
+						if ((tmp < 0) || (!_superblocktype_2_3 && !tmp))
+							_toneLevel[ch][sb][i] = 0;
+						else
+							_toneLevel[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
+					}
+				}
+			} else {
+				if (sb > 4) {
+					for (ch = 0; ch < _channels; ch++) {
+						for (i = 0; i < 64; i++) {
+							tmp = _toneLevelIdxBase[ch][sb][i / 8] -
+							      _toneLevelIdxHi1[ch][2][i / 8][i % 8] -
+							      _toneLevelIdxHi2[ch][sb - 4];
+							_toneLevelIdx[ch][sb][i] = tmp & 0xff;
+							if ((tmp < 0) || (!_superblocktype_2_3 && !tmp))
+								_toneLevel[ch][sb][i] = 0;
+							else
+								_toneLevel[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
+						}
+					}
+				} else {
+					for (ch = 0; ch < _channels; ch++) {
+						for (i = 0; i < 64; i++) {
+							tmp = _toneLevelIdx[ch][sb][i] = _toneLevelIdxBase[ch][sb][i / 8];
+							if ((tmp < 0) || (!_superblocktype_2_3 && !tmp))
+								_toneLevel[ch][sb][i] = 0;
+							else
+								_toneLevel[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
+						}
+					}
+				}
+			}
+		}
+	}
+}
+
+/**
+ * Related to synthesis filter
+ * Called by process_subpacket_11
+ * c is built with data from subpacket 11
+ * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples
+ *
+ * @param tone_level_idx
+ * @param tone_level_idx_temp
+ * @param coding_method        q->coding_method[0][0][0]
+ * @param nb_channels          number of channels
+ * @param c                    coming from subpacket 11, passed as 8*c
+ * @param superblocktype_2_3   flag based on superblock packet type
+ * @param cm_table_select      q->cm_table_select
+ */
+void QDM2Stream::fill_coding_method_array(sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
+                sb_int8_array coding_method, int nb_channels,
+                int c, int superblocktype_2_3, int cm_table_select) {
+	int ch, sb, j;
+	int tmp, acc, esp_40, comp;
+	int add1, add2, add3, add4;
+	// TODO : Remove multres 64 bit variable necessity...
+	int64_t multres;
+
+	// This should never happen
+	if (nb_channels <= 0)
+		return;
+	if (!superblocktype_2_3) {
+		warning("QDM2 This case is untested, no samples available");
+		for (ch = 0; ch < nb_channels; ch++)
+			for (sb = 0; sb < 30; sb++) {
+				for (j = 1; j < 63; j++) {  // The loop only iterates to 63 so the code doesn't overflow the buffer
+					add1 = tone_level_idx[ch][sb][j] - 10;
+					if (add1 < 0)
+						add1 = 0;
+					add2 = add3 = add4 = 0;
+					if (sb > 1) {
+						add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
+						if (add2 < 0)
+							add2 = 0;
+					}
+					if (sb > 0) {
+						add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
+						if (add3 < 0)
+							add3 = 0;
+					}
+					if (sb < 29) {
+						add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
+						if (add4 < 0)
+							add4 = 0;
+					}
+					tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
+					if (tmp < 0)
+						tmp = 0;
+					tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
+				}
+				tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
+			}
+			acc = 0;
+			for (ch = 0; ch < nb_channels; ch++)
+				for (sb = 0; sb < 30; sb++)
+					for (j = 0; j < 64; j++)
+						acc += tone_level_idx_temp[ch][sb][j];
+
+			multres = 0x66666667 * (acc * 10);
+			esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
+			for (ch = 0;  ch < nb_channels; ch++)
+				for (sb = 0; sb < 30; sb++)
+					for (j = 0; j < 64; j++) {
+						comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
+						if (comp < 0)
+							comp += 0xff;
+						comp /= 256; // signed shift
+						switch(sb) {
+							case 0:
+								if (comp < 30)
+									comp = 30;
+								comp += 15;
+								break;
+							case 1:
+								if (comp < 24)
+									comp = 24;
+								comp += 10;
+								break;
+							case 2:
+							case 3:
+							case 4:
+								if (comp < 16)
+									comp = 16;
+						}
+						if (comp <= 5)
+							tmp = 0;
+						else if (comp <= 10)
+							tmp = 10;
+						else if (comp <= 16)
+							tmp = 16;
+						else if (comp <= 24)
+							tmp = -1;
+						else
+							tmp = 0;
+						coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
+					}
+			for (sb = 0; sb < 30; sb++)
+				fix_coding_method_array(sb, nb_channels, coding_method);
+			for (ch = 0; ch < nb_channels; ch++)
+				for (sb = 0; sb < 30; sb++)
+					for (j = 0; j < 64; j++)
+						if (sb >= 10) {
+							if (coding_method[ch][sb][j] < 10)
+								coding_method[ch][sb][j] = 10;
+						} else {
+							if (sb >= 2) {
+								if (coding_method[ch][sb][j] < 16)
+									coding_method[ch][sb][j] = 16;
+							} else {
+								if (coding_method[ch][sb][j] < 30)
+									coding_method[ch][sb][j] = 30;
+							}
+						}
+	} else { // superblocktype_2_3 != 0
+		for (ch = 0; ch < nb_channels; ch++)
+			for (sb = 0; sb < 30; sb++)
+				for (j = 0; j < 64; j++)
+					coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
+	}
+}
+
+/**
+ *
+ * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8
+ * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used
+ *
+ * @param gb        bitreader context
+ * @param length    packet length in bits
+ * @param sb_min    lower subband processed (sb_min included)
+ * @param sb_max    higher subband processed (sb_max excluded)
+ */
+void QDM2Stream::synthfilt_build_sb_samples(GetBitContext *gb, int length, int sb_min, int sb_max) {
+	int sb, j, k, n, ch, run, channels;
+	int joined_stereo, zero_encoding, chs;
+	int type34_first;
+	float type34_div = 0;
+	float type34_predictor;
+	float samples[10], sign_bits[16];
+
+	if (length == 0) {
+		// If no data use noise
+		for (sb = sb_min; sb < sb_max; sb++)
+			build_sb_samples_from_noise(sb);
+
+		return;
+	}
+
+	for (sb = sb_min; sb < sb_max; sb++) {
+		FIX_NOISE_IDX(_noiseIdx);
+
+		channels = _channels;
+
+		if (_channels <= 1 || sb < 12)
+			joined_stereo = 0;
+		else if (sb >= 24)
+			joined_stereo = 1;
+		else
+			joined_stereo = (BITS_LEFT(length,gb) >= 1) ? getBits1 (gb) : 0;
+
+		if (joined_stereo) {
+			if (BITS_LEFT(length,gb) >= 16)
+				for (j = 0; j < 16; j++)
+					sign_bits[j] = getBits1(gb);
+
+			for (j = 0; j < 64; j++)
+				if (_codingMethod[1][sb][j] > _codingMethod[0][sb][j])
+					_codingMethod[0][sb][j] = _codingMethod[1][sb][j];
+
+			fix_coding_method_array(sb, _channels, _codingMethod);
+			channels = 1;
+		}
+
+		for (ch = 0; ch < channels; ch++) {
+			zero_encoding = (BITS_LEFT(length,gb) >= 1) ? getBits1(gb) : 0;
+			type34_predictor = 0.0;
+			type34_first = 1;
+
+			for (j = 0; j < 128; ) {
+				switch (_codingMethod[ch][sb][j / 2]) {
+					case 8:
+						if (BITS_LEFT(length,gb) >= 10) {
+							if (zero_encoding) {
+								for (k = 0; k < 5; k++) {
+								if ((j + 2 * k) >= 128)
+									break;
+									samples[2 * k] = getBits1(gb) ? dequant_1bit[joined_stereo][2 * getBits1(gb)] : 0;
+								}
+							} else {
+								n = getBits(gb, 8);
+								for (k = 0; k < 5; k++)
+									samples[2 * k] = dequant_1bit[joined_stereo][_randomDequantIndex[n][k]];
+							}
+							for (k = 0; k < 5; k++)
+								samples[2 * k + 1] = SB_DITHERING_NOISE(sb, _noiseIdx);
+						} else {
+							for (k = 0; k < 10; k++)
+								samples[k] = SB_DITHERING_NOISE(sb, _noiseIdx);
+						}
+						run = 10;
+						break;
+
+					case 10:
+						if (BITS_LEFT(length,gb) >= 1) {
+							double f = 0.81;
+
+							if (getBits1(gb))
+								f = -f;
+							f -= _noiseSamples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
+							samples[0] = f;
+						} else {
+							samples[0] = SB_DITHERING_NOISE(sb, _noiseIdx);
+						}
+						run = 1;
+						break;
+
+					case 16:
+						if (BITS_LEFT(length,gb) >= 10) {
+							if (zero_encoding) {
+								for (k = 0; k < 5; k++) {
+									if ((j + k) >= 128)
+										break;
+									samples[k] = (getBits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * getBits1(gb)];
+								}
+							} else {
+								n = getBits (gb, 8);
+								for (k = 0; k < 5; k++)
+									samples[k] = dequant_1bit[joined_stereo][_randomDequantIndex[n][k]];
+							}
+						} else {
+							for (k = 0; k < 5; k++)
+								samples[k] = SB_DITHERING_NOISE(sb, _noiseIdx);
+						}
+						run = 5;
+						break;
+
+					case 24:
+						if (BITS_LEFT(length,gb) >= 7) {
+							n = getBits(gb, 7);
+							for (k = 0; k < 3; k++)
+								samples[k] = (_randomDequantType24[n][k] - 2.0) * 0.5;
+						} else {
+							for (k = 0; k < 3; k++)
+								samples[k] = SB_DITHERING_NOISE(sb, _noiseIdx);
+						}
+						run = 3;
+						break;
+
+					case 30:
+						if (BITS_LEFT(length,gb) >= 4)
+							samples[0] = type30_dequant[qdm2_get_vlc(gb, &_vlcTabType30, 0, 1)];
+						else
+							samples[0] = SB_DITHERING_NOISE(sb, _noiseIdx);
+
+						run = 1;
+						break;
+
+					case 34:
+						if (BITS_LEFT(length,gb) >= 7) {
+							if (type34_first) {
+								type34_div = (float)(1 << getBits(gb, 2));
+								samples[0] = ((float)getBits(gb, 5) - 16.0) / 15.0;
+								type34_predictor = samples[0];
+								type34_first = 0;
+							} else {
+								samples[0] = type34_delta[qdm2_get_vlc(gb, &_vlcTabType34, 0, 1)] / type34_div + type34_predictor;
+								type34_predictor = samples[0];
+							}
+						} else {
+							samples[0] = SB_DITHERING_NOISE(sb, _noiseIdx);
+						}
+						run = 1;
+						break;
+
+					default:
+						samples[0] = SB_DITHERING_NOISE(sb, _noiseIdx);
+						run = 1;
+						break;
+				}
+
+				if (joined_stereo) {
+					float tmp[10][MPA_MAX_CHANNELS];
+
+					for (k = 0; k < run; k++) {
+						tmp[k][0] = samples[k];
+						tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
+					}
+					for (chs = 0; chs < _channels; chs++)
+						for (k = 0; k < run; k++)
+							if ((j + k) < 128)
+								_sbSamples[chs][j + k][sb] = (int32)(_toneLevel[chs][sb][((j + k)/2)] * tmp[k][chs] + .5);
+				} else {
+					for (k = 0; k < run; k++)
+						if ((j + k) < 128)
+							_sbSamples[ch][j + k][sb] = (int32)(_toneLevel[ch][sb][(j + k)/2] * samples[k] + .5);
+				}
+
+				j += run;
+			} // j loop
+		} // channel loop
+	} // subband loop
+}
+
+/**
+ * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]).
+ * This is similar to process_subpacket_9, but for a single channel and for element [0]
+ * same VLC tables as process_subpacket_9 are used.
+ *
+ * @param quantized_coeffs    pointer to quantized_coeffs[ch][0]
+ * @param gb        bitreader context
+ * @param length    packet length in bits
+ */
+void QDM2Stream::init_quantized_coeffs_elem0(int8 *quantized_coeffs, GetBitContext *gb, int length) {
+	int i, k, run, level, diff;
+
+	if (BITS_LEFT(length,gb) < 16)
+		return;
+	level = qdm2_get_vlc(gb, &_vlcTabLevel, 0, 2);
+
+	quantized_coeffs[0] = level;
+
+	for (i = 0; i < 7; ) {
+		if (BITS_LEFT(length,gb) < 16)
+			break;
+		run = qdm2_get_vlc(gb, &_vlcTabRun, 0, 1) + 1;
+
+		if (BITS_LEFT(length,gb) < 16)
+			break;
+		diff = qdm2_get_se_vlc(&_vlcTabDiff, gb, 2);
+
+		for (k = 1; k <= run; k++)
+			quantized_coeffs[i + k] = (level + ((k * diff) / run));
+
+		level += diff;
+		i += run;
+	}
+}
+
+/**
+ * Related to synthesis filter, process data from packet 10
+ * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
+ * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10
+ *
+ * @param gb        bitreader context
+ * @param length    packet length in bits
+ */
+void QDM2Stream::init_tone_level_dequantization(GetBitContext *gb, int length) {
+	int sb, j, k, n, ch;
+
+	for (ch = 0; ch < _channels; ch++) {
+		init_quantized_coeffs_elem0(_quantizedCoeffs[ch][0], gb, length);
+
+		if (BITS_LEFT(length,gb) < 16) {
+			memset(_quantizedCoeffs[ch][0], 0, 8);
+			break;
+		}
+	}
+
+	n = _subSampling + 1;
+
+	for (sb = 0; sb < n; sb++)
+		for (ch = 0; ch < _channels; ch++)
+			for (j = 0; j < 8; j++) {
+				if (BITS_LEFT(length,gb) < 1)
+					break;
+				if (getBits1(gb)) {
+					for (k=0; k < 8; k++) {
+						if (BITS_LEFT(length,gb) < 16)
+							break;
+						_toneLevelIdxHi1[ch][sb][j][k] = qdm2_get_vlc(gb, &_vlcTabToneLevelIdxHi1, 0, 2);
+					}
+				} else {
+					for (k=0; k < 8; k++)
+						_toneLevelIdxHi1[ch][sb][j][k] = 0;
+				}
+			}
+
+	n = QDM2_SB_USED(_subSampling) - 4;
+
+	for (sb = 0; sb < n; sb++)
+		for (ch = 0; ch < _channels; ch++) {
+			if (BITS_LEFT(length,gb) < 16)
+				break;
+			_toneLevelIdxHi2[ch][sb] = qdm2_get_vlc(gb, &_vlcTabToneLevelIdxHi2, 0, 2);
+			if (sb > 19)
+				_toneLevelIdxHi2[ch][sb] -= 16;
+			else
+				for (j = 0; j < 8; j++)
+					_toneLevelIdxMid[ch][sb][j] = -16;
+		}
+
+	n = QDM2_SB_USED(_subSampling) - 5;
+
+	for (sb = 0; sb < n; sb++) {
+		for (ch = 0; ch < _channels; ch++) {
+			for (j = 0; j < 8; j++) {
+				if (BITS_LEFT(length,gb) < 16)
+					break;
+				_toneLevelIdxMid[ch][sb][j] = qdm2_get_vlc(gb, &_vlcTabToneLevelIdxMid, 0, 2) - 32;
+			}
+		}
+	}
+}
+
+/**
+ * Process subpacket 9, init quantized_coeffs with data from it
+ *
+ * @param node    pointer to node with packet
+ */
+void QDM2Stream::process_subpacket_9(QDM2SubPNode *node) {
+	GetBitContext gb;
+	int i, j, k, n, ch, run, level, diff;
+
+	initGetBits(&gb, node->packet->data, node->packet->size*8);
+
+	n = coeff_per_sb_for_avg[_coeffPerSbSelect][QDM2_SB_USED(_subSampling) - 1] + 1; // same as averagesomething function
+
+	for (i = 1; i < n; i++)
+		for (ch = 0; ch < _channels; ch++) {
+			level = qdm2_get_vlc(&gb, &_vlcTabLevel, 0, 2);
+			_quantizedCoeffs[ch][i][0] = level;
+
+			for (j = 0; j < (8 - 1); ) {
+				run = qdm2_get_vlc(&gb, &_vlcTabRun, 0, 1) + 1;
+				diff = qdm2_get_se_vlc(&_vlcTabDiff, &gb, 2);
+
+				for (k = 1; k <= run; k++)
+					_quantizedCoeffs[ch][i][j + k] = (level + ((k*diff) / run));
+
+				level += diff;
+				j += run;
+			}
+		}
+
+	for (ch = 0; ch < _channels; ch++)
+		for (i = 0; i < 8; i++)
+			_quantizedCoeffs[ch][0][i] = 0;
+}
+
+/**
+ * Process subpacket 10 if not null, else
+ *
+ * @param node      pointer to node with packet
+ * @param length    packet length in bits
+ */
+void QDM2Stream::process_subpacket_10(QDM2SubPNode *node, int length) {
+	GetBitContext gb;
+
+	initGetBits(&gb, ((node == NULL) ? _emptyBuffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
+
+	if (length != 0) {
+		init_tone_level_dequantization(&gb, length);
+		fill_tone_level_array(1);
+	} else {
+		fill_tone_level_array(0);
+	}
+}
+
+/**
+ * Process subpacket 11
+ *
+ * @param node      pointer to node with packet
+ * @param length    packet length in bit
+ */
+void QDM2Stream::process_subpacket_11(QDM2SubPNode *node, int length) {
+	GetBitContext gb;
+
+	initGetBits(&gb, ((node == NULL) ? _emptyBuffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
+	if (length >= 32) {
+		int c = getBits (&gb, 13);
+
+		if (c > 3)
+			fill_coding_method_array(_toneLevelIdx, _toneLevelIdxTemp, _codingMethod,
+			                         _channels, 8*c, _superblocktype_2_3, _cmTableSelect);
+	}
+
+	synthfilt_build_sb_samples(&gb, length, 0, 8);
+}
+
+/**
+ * Process subpacket 12
+ *
+ * @param node      pointer to node with packet
+ * @param length    packet length in bits
+ */
+void QDM2Stream::process_subpacket_12(QDM2SubPNode *node, int length) {
+	GetBitContext gb;
+
+	initGetBits(&gb, ((node == NULL) ? _emptyBuffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
+	synthfilt_build_sb_samples(&gb, length, 8, QDM2_SB_USED(_subSampling));
+}
+
+/*
+ * Process new subpackets for synthesis filter
+ *
+ * @param list    list with synthesis filter packets (list D)
+ */
+void QDM2Stream::process_synthesis_subpackets(QDM2SubPNode *list) {
+	struct QDM2SubPNode *nodes[4];
+
+	nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
+	if (nodes[0] != NULL)
+		process_subpacket_9(nodes[0]);
+
+	nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
+	if (nodes[1] != NULL)
+		process_subpacket_10(nodes[1], nodes[1]->packet->size << 3);
+	else
+		process_subpacket_10(NULL, 0);
+
+	nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
+	if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
+		process_subpacket_11(nodes[2], (nodes[2]->packet->size << 3));
+	else
+		process_subpacket_11(NULL, 0);
+
+	nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
+	if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
+		process_subpacket_12(nodes[3], (nodes[3]->packet->size << 3));
+	else
+		process_subpacket_12(NULL, 0);
+}
+
+/*
+ * Decode superblock, fill packet lists.
+ *
+ */
+void QDM2Stream::qdm2_decode_super_block(void) {
+	GetBitContext gb;
+	struct QDM2SubPacket header, *packet;
+	int i, packet_bytes, sub_packet_size, subPacketsD;
+	unsigned int next_index = 0;
+
+	memset(_toneLevelIdxHi1, 0, sizeof(_toneLevelIdxHi1));
+	memset(_toneLevelIdxMid, 0, sizeof(_toneLevelIdxMid));
+	memset(_toneLevelIdxHi2, 0, sizeof(_toneLevelIdxHi2));
+
+	_subPacketsB = 0;
+	subPacketsD = 0;
+
+	average_quantized_coeffs(); // average elements in quantized_coeffs[max_ch][10][8]
+
+	initGetBits(&gb, _compressedData, _packetSize*8);
+	qdm2_decode_sub_packet_header(&gb, &header);
+
+	if (header.type < 2 || header.type >= 8) {
+		_hasErrors = true;
+		error("QDM2 : bad superblock type");
+		return;
+	}
+
+	_superblocktype_2_3 = (header.type == 2 || header.type == 3);
+	packet_bytes = (_packetSize - getBitsCount(&gb) / 8);
+
+	initGetBits(&gb, header.data, header.size*8);
+
+	if (header.type == 2 || header.type == 4 || header.type == 5) {
+		int csum = 257 * getBits(&gb, 8) + 2 * getBits(&gb, 8);
+
+		csum = qdm2_packet_checksum(_compressedData, _packetSize, csum);
+
+		if (csum != 0) {
+			_hasErrors = true;
+			error("QDM2 : bad packet checksum");
+			return;
+		}
+	}
+
+	_subPacketListB[0].packet = NULL;
+	_subPacketListD[0].packet = NULL;
+
+	for (i = 0; i < 6; i++)
+		if (--_fftLevelExp[i] < 0)
+			_fftLevelExp[i] = 0;
+
+	for (i = 0; packet_bytes > 0; i++) {
+		int j;
+
+		_subPacketListA[i].next = NULL;
+
+		if (i > 0) {
+			_subPacketListA[i - 1].next = &_subPacketListA[i];
+
+			// seek to next block
+			initGetBits(&gb, header.data, header.size*8);
+			skipBits(&gb, next_index*8);
+
+			if (next_index >= header.size)
+				break;
+		}
+
+		// decode subpacket
+		packet = &_subPackets[i];
+		qdm2_decode_sub_packet_header(&gb, packet);
+		next_index = packet->size + getBitsCount(&gb) / 8;
+		sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
+
+		if (packet->type == 0)
+			break;
+
+		if (sub_packet_size > packet_bytes) {
+			if (packet->type != 10 && packet->type != 11 && packet->type != 12)
+				break;
+			packet->size += packet_bytes - sub_packet_size;
+		}
+
+		packet_bytes -= sub_packet_size;
+
+		// add subpacket to 'all subpackets' list
+		_subPacketListA[i].packet = packet;
+
+		// add subpacket to related list
+		if (packet->type == 8) {
+			error("Unsupported packet type 8");
+			return;
+		} else if (packet->type >= 9 && packet->type <= 12) {
+			// packets for MPEG Audio like Synthesis Filter
+			QDM2_LIST_ADD(_subPacketListD, subPacketsD, packet);
+		} else if (packet->type == 13) {
+			for (j = 0; j < 6; j++)
+				_fftLevelExp[j] = getBits(&gb, 6);
+		} else if (packet->type == 14) {
+			for (j = 0; j < 6; j++)
+				_fftLevelExp[j] = qdm2_get_vlc(&gb, &_fftLevelExpVlc, 0, 2);
+		} else if (packet->type == 15) {
+			error("Unsupported packet type 15");
+			return;
+		} else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
+			// packets for FFT
+			QDM2_LIST_ADD(_subPacketListB, _subPacketsB, packet);
+		}
+	} // Packet bytes loop
+
+// ****************************************************************
+	if (_subPacketListD[0].packet != NULL) {
+		process_synthesis_subpackets(_subPacketListD);
+		_doSynthFilter = 1;
+	} else if (_doSynthFilter) {
+		process_subpacket_10(NULL, 0);
+		process_subpacket_11(NULL, 0);
+		process_subpacket_12(NULL, 0);
+	}
+// ****************************************************************
+}
+
+void QDM2Stream::qdm2_fft_init_coefficient(int sub_packet, int offset, int duration,
+                                           int channel, int exp, int phase) {
+	if (_fftCoefsMinIndex[duration] < 0)
+	    _fftCoefsMinIndex[duration] = _fftCoefsIndex;
+
+	_fftCoefs[_fftCoefsIndex].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
+	_fftCoefs[_fftCoefsIndex].channel = channel;
+	_fftCoefs[_fftCoefsIndex].offset = offset;
+	_fftCoefs[_fftCoefsIndex].exp = exp;
+	_fftCoefs[_fftCoefsIndex].phase = phase;
+	_fftCoefsIndex++;
+}
+
+void QDM2Stream::qdm2_fft_decode_tones(int duration, GetBitContext *gb, int b) {
+	int channel, stereo, phase, exp;
+	int local_int_4,  local_int_8,  stereo_phase,  local_int_10;
+	int local_int_14, stereo_exp, local_int_20, local_int_28;
+	int n, offset;
+
+	local_int_4 = 0;
+	local_int_28 = 0;
+	local_int_20 = 2;
+	local_int_8 = (4 - duration);
+	local_int_10 = 1 << (_groupOrder - duration - 1);
+	offset = 1;
+
+	while (1) {
+		if (_superblocktype_2_3) {
+			while ((n = qdm2_get_vlc(gb, &_vlcTabFftToneOffset[local_int_8], 1, 2)) < 2) {
+				offset = 1;
+				if (n == 0) {
+					local_int_4 += local_int_10;
+					local_int_28 += (1 << local_int_8);
+				} else {
+					local_int_4 += 8*local_int_10;
+					local_int_28 += (8 << local_int_8);
+				}
+			}
+			offset += (n - 2);
+		} else {
+			offset += qdm2_get_vlc(gb, &_vlcTabFftToneOffset[local_int_8], 1, 2);
+			while (offset >= (local_int_10 - 1)) {
+				offset += (1 - (local_int_10 - 1));
+				local_int_4  += local_int_10;
+				local_int_28 += (1 << local_int_8);
+			}
+		}
+
+		if (local_int_4 >= _blockSize)
+			return;
+
+		local_int_14 = (offset >> local_int_8);
+
+		if (_channels > 1) {
+			channel = getBits1(gb);
+			stereo = getBits1(gb);
+		} else {
+			channel = 0;
+			stereo = 0;
+		}
+
+		exp = qdm2_get_vlc(gb, (b ? &_fftLevelExpVlc : &_fftLevelExpAltVlc), 0, 2);
+		exp += _fftLevelExp[fft_level_index_table[local_int_14]];
+		exp = (exp < 0) ? 0 : exp;
+
+		phase = getBits(gb, 3);
+		stereo_exp = 0;
+		stereo_phase = 0;
+
+		if (stereo) {
+			stereo_exp = (exp - qdm2_get_vlc(gb, &_fftStereoExpVlc, 0, 1));
+			stereo_phase = (phase - qdm2_get_vlc(gb, &_fftStereoPhaseVlc, 0, 1));
+			if (stereo_phase < 0)
+				stereo_phase += 8;
+		}
+
+		if (_frequencyRange > (local_int_14 + 1)) {
+			int sub_packet = (local_int_20 + local_int_28);
+
+			qdm2_fft_init_coefficient(sub_packet, offset, duration, channel, exp, phase);
+			if (stereo)
+				qdm2_fft_init_coefficient(sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
+		}
+
+		offset++;
+	}
+}
+
+void QDM2Stream::qdm2_decode_fft_packets(void) {
+	int i, j, min, max, value, type, unknown_flag;
+	GetBitContext gb;
+
+	if (_subPacketListB[0].packet == NULL)
+		return;
+
+	// reset minimum indexes for FFT coefficients
+	_fftCoefsIndex = 0;
+	for (i=0; i < 5; i++)
+		_fftCoefsMinIndex[i] = -1;
+
+	// process subpackets ordered by type, largest type first
+	for (i = 0, max = 256; i < _subPacketsB; i++) {
+		QDM2SubPacket *packet= NULL;
+
+		// find subpacket with largest type less than max
+		for (j = 0, min = 0; j < _subPacketsB; j++) {
+			value = _subPacketListB[j].packet->type;
+			if (value > min && value < max) {
+				min = value;
+				packet = _subPacketListB[j].packet;
+			}
+		}
+
+		max = min;
+
+		// check for errors (?)
+		if (!packet)
+			return;
+
+		if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
+			return;
+
+		// decode FFT tones
+		initGetBits(&gb, packet->data, packet->size*8);
+
+		if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
+			unknown_flag = 1;
+		else
+			unknown_flag = 0;
+
+		type = packet->type;
+
+		if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
+			int duration = _subSampling + 5 - (type & 15);
+
+			if (duration >= 0 && duration < 4) { // TODO: Should be <= 4?
+				qdm2_fft_decode_tones(duration, &gb, unknown_flag);
+			}
+		} else if (type == 31) {
+			for (j=0; j < 4; j++) {
+				qdm2_fft_decode_tones(j, &gb, unknown_flag);
+			}
+		} else if (type == 46) {
+			for (j=0; j < 6; j++)
+				_fftLevelExp[j] = getBits(&gb, 6);
+			for (j=0; j < 4; j++) {
+				qdm2_fft_decode_tones(j, &gb, unknown_flag);
+			}
+		}
+	} // Loop on B packets
+
+	// calculate maximum indexes for FFT coefficients
+	for (i = 0, j = -1; i < 5; i++)
+		if (_fftCoefsMinIndex[i] >= 0) {
+			if (j >= 0)
+				_fftCoefsMaxIndex[j] = _fftCoefsMinIndex[i];
+			j = i;
+		}
+	if (j >= 0)
+		_fftCoefsMaxIndex[j] = _fftCoefsIndex;
+}
+
+void QDM2Stream::qdm2_fft_generate_tone(FFTTone *tone)
+{
+	float level, f[6];
+	int i;
+	QDM2Complex c;
+	const double iscale = 2.0 * M_PI / 512.0;
+
+	tone->phase += tone->phase_shift;
+
+	// calculate current level (maximum amplitude) of tone
+	level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
+	c.im = level * sin(tone->phase*iscale);
+	c.re = level * cos(tone->phase*iscale);
+
+	// generate FFT coefficients for tone
+	if (tone->duration >= 3 || tone->cutoff >= 3) {
+	    tone->complex[0].im += c.im;
+	    tone->complex[0].re += c.re;
+	    tone->complex[1].im -= c.im;
+	    tone->complex[1].re -= c.re;
+	} else {
+		f[1] = -tone->table[4];
+		f[0] =  tone->table[3] - tone->table[0];
+		f[2] =  1.0 - tone->table[2] - tone->table[3];
+		f[3] =  tone->table[1] + tone->table[4] - 1.0;
+		f[4] =  tone->table[0] - tone->table[1];
+		f[5] =  tone->table[2];
+		for (i = 0; i < 2; i++) {
+			tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i];
+			tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
+		}
+		for (i = 0; i < 4; i++) {
+			tone->complex[i].re += c.re * f[i+2];
+			tone->complex[i].im += c.im * f[i+2];
+		}
+	}
+
+	// copy the tone if it has not yet died out
+	if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
+		memcpy(&_fftTones[_fftToneEnd], tone, sizeof(FFTTone));
+		_fftToneEnd = (_fftToneEnd + 1) % 1000;
+	}
+}
+
+void QDM2Stream::qdm2_fft_tone_synthesizer(uint8 sub_packet) {
+	int i, j, ch;
+	const double iscale = 0.25 * M_PI;
+
+	for (ch = 0; ch < _channels; ch++) {
+		memset(_fft.complex[ch], 0, _frameSize * sizeof(QDM2Complex));
+	}
+
+	// apply FFT tones with duration 4 (1 FFT period)
+	if (_fftCoefsMinIndex[4] >= 0)
+		for (i = _fftCoefsMinIndex[4]; i < _fftCoefsMaxIndex[4]; i++) {
+			float level;
+			QDM2Complex c;
+
+			if (_fftCoefs[i].sub_packet != sub_packet)
+				break;
+
+			ch = (_channels == 1) ? 0 : _fftCoefs[i].channel;
+			level = (_fftCoefs[i].exp < 0) ? 0.0 : fft_tone_level_table[_superblocktype_2_3 ? 0 : 1][_fftCoefs[i].exp & 63];
+
+			c.re = level * cos(_fftCoefs[i].phase * iscale);
+			c.im = level * sin(_fftCoefs[i].phase * iscale);
+			_fft.complex[ch][_fftCoefs[i].offset + 0].re += c.re;
+			_fft.complex[ch][_fftCoefs[i].offset + 0].im += c.im;
+			_fft.complex[ch][_fftCoefs[i].offset + 1].re -= c.re;
+			_fft.complex[ch][_fftCoefs[i].offset + 1].im -= c.im;
+		}
+
+	// generate existing FFT tones
+	for (i = _fftToneEnd; i != _fftToneStart; ) {
+		qdm2_fft_generate_tone(&_fftTones[_fftToneStart]);
+		_fftToneStart = (_fftToneStart + 1) % 1000;
+	}
+
+	// create and generate new FFT tones with duration 0 (long) to 3 (short)
+	for (i = 0; i < 4; i++)
+		if (_fftCoefsMinIndex[i] >= 0) {
+			for (j = _fftCoefsMinIndex[i]; j < _fftCoefsMaxIndex[i]; j++) {
+				int offset, four_i;
+				FFTTone tone;
+
+				if (_fftCoefs[j].sub_packet != sub_packet)
+					break;
+
+				four_i = (4 - i);
+				offset = _fftCoefs[j].offset >> four_i;
+				ch = (_channels == 1) ? 0 : _fftCoefs[j].channel;
+
+				if (offset < _frequencyRange) {
+					if (offset < 2)
+						tone.cutoff = offset;
+					else
+						tone.cutoff = (offset >= 60) ? 3 : 2;
+
+					tone.level = (_fftCoefs[j].exp < 0) ? 0.0 : fft_tone_level_table[_superblocktype_2_3 ? 0 : 1][_fftCoefs[j].exp & 63];
+					tone.complex = &_fft.complex[ch][offset];
+					tone.table = fft_tone_sample_table[i][_fftCoefs[j].offset - (offset << four_i)];
+					tone.phase = 64 * _fftCoefs[j].phase - (offset << 8) - 128;
+					tone.phase_shift = (2 * _fftCoefs[j].offset + 1) << (7 - four_i);
+					tone.duration = i;
+					tone.time_index = 0;
+
+					qdm2_fft_generate_tone(&tone);
+				}
+			}
+			_fftCoefsMinIndex[i] = j;
+		}
+}
+
+void QDM2Stream::qdm2_calculate_fft(int channel) {
+	int i;
+
+	_fft.complex[channel][0].re *= 2.0f;
+	_fft.complex[channel][0].im = 0.0f;
+
+	rdftCalc(&_rdftCtx, (float *)_fft.complex[channel]);
+
+	// add samples to output buffer
+	for (i = 0; i < ((_fftFrameSize + 15) & ~15); i++)
+		_outputBuffer[_channels * i + channel] += ((float *) _fft.complex[channel])[i];
+}
+
+/**
+ * @param index    subpacket number
+ */
+void QDM2Stream::qdm2_synthesis_filter(uint8 index)
+{
+	int16 samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
+	int i, k, ch, sb_used, sub_sampling, dither_state = 0;
+
+	// copy sb_samples
+	sb_used = QDM2_SB_USED(_subSampling);
+
+	for (ch = 0; ch < _channels; ch++)
+		for (i = 0; i < 8; i++)
+			for (k = sb_used; k < 32; k++)
+				_sbSamples[ch][(8 * index) + i][k] = 0;
+
+	for (ch = 0; ch < _channels; ch++) {
+		int16 *samples_ptr = samples + ch;
+
+		for (i = 0; i < 8; i++) {
+			ff_mpa_synth_filter(_synthBuf[ch], &(_synthBufOffset[ch]),
+			                    ff_mpa_synth_window, &dither_state,
+			                    samples_ptr, _channels,
+			                    _sbSamples[ch][(8 * index) + i]);
+			samples_ptr += 32 * _channels;
+		}
+	}
+
+	// add samples to output buffer
+	sub_sampling = (4 >> _subSampling);
+
+	for (ch = 0; ch < _channels; ch++)
+		for (i = 0; i < _sFrameSize; i++)
+			_outputBuffer[_channels * i + ch] += (float)(samples[_channels * sub_sampling * i + ch] >> (sizeof(int16)*8-16));
+}
+
+int QDM2Stream::qdm2_decodeFrame(Common::SeekableReadStream *in) {
+	debug(1, "QDM2Stream::qdm2_decodeFrame in->pos(): %d in->size(): %d", in->pos(), in->size());
+	int ch, i;
+	const int frame_size = (_sFrameSize * _channels);
+
+	// If we're in any packet but the first, seek back to the first
+	if (_subPacket == 0)
+		_superBlockStart = in->pos();
+	else
+		in->seek(_superBlockStart);
+
+	// select input buffer
+	if (in->eos() || in->pos() >= in->size()) {
+		debug(1, "QDM2Stream::qdm2_decodeFrame End of Input Stream");
+		return 0;
+	}
+
+	if ((in->size() - in->pos()) < _packetSize) {
+		debug(1, "QDM2Stream::qdm2_decodeFrame Insufficient Packet Data in Input Stream Found: %d Need: %d", in->size() - in->pos(), _packetSize);
+		return 0;
+	}
+
+	if (!in->eos()) {
+		in->read(_compressedData, _packetSize);
+		debug(1, "QDM2Stream::qdm2_decodeFrame constructed input data");
+	}
+
+	// copy old block, clear new block of output samples
+	memmove(_outputBuffer, &_outputBuffer[frame_size], frame_size * sizeof(float));
+	memset(&_outputBuffer[frame_size], 0, frame_size * sizeof(float));
+	debug(1, "QDM2Stream::qdm2_decodeFrame cleared outputBuffer");
+
+	if (!in->eos()) {
+		// decode block of QDM2 compressed data
+		debug(1, "QDM2Stream::qdm2_decodeFrame decode block of QDM2 compressed data");
+		if (_subPacket == 0) {
+			_hasErrors = false; // reset it for a new super block
+			debug(1, "QDM2 : Superblock follows");
+			qdm2_decode_super_block();
+		}
+
+		// parse subpackets
+		debug(1, "QDM2Stream::qdm2_decodeFrame parse subpackets");
+		if (!_hasErrors) {
+			if (_subPacket == 2) {
+				debug(1, "QDM2Stream::qdm2_decodeFrame qdm2_decode_fft_packets()");
+				qdm2_decode_fft_packets();
+			}
+
+			debug(1, "QDM2Stream::qdm2_decodeFrame qdm2_fft_tone_synthesizer(%d)", _subPacket);
+			qdm2_fft_tone_synthesizer(_subPacket);
+		}
+
+		// sound synthesis stage 1 (FFT)
+		debug(1, "QDM2Stream::qdm2_decodeFrame sound synthesis stage 1 (FFT)");
+		for (ch = 0; ch < _channels; ch++) {
+			qdm2_calculate_fft(ch);
+
+			if (!_hasErrors && _subPacketListC[0].packet != NULL) {
+				error("QDM2 : has errors, and C list is not empty");
+				return 0;
+			}
+		}
+
+		// sound synthesis stage 2 (MPEG audio like synthesis filter)
+		debug(1, "QDM2Stream::qdm2_decodeFrame sound synthesis stage 2 (MPEG audio like synthesis filter)");
+		if (!_hasErrors && _doSynthFilter)
+			qdm2_synthesis_filter(_subPacket);
+
+		_subPacket = (_subPacket + 1) % 16;
+
+		if(_hasErrors)
+			warning("QDM2 Packet error...");
+
+		// clip and convert output float[] to 16bit signed samples
+		debug(1, "QDM2Stream::qdm2_decodeFrame clip and convert output float[] to 16bit signed samples");
+	}
+
+	for (i = 0; i < frame_size; i++) {
+		int value = (int)_outputBuffer[i];
+
+		if (value > SOFTCLIP_THRESHOLD)
+			value = (value >  HARDCLIP_THRESHOLD) ?  32767 :  _softclipTable[ value - SOFTCLIP_THRESHOLD];
+		else if (value < -SOFTCLIP_THRESHOLD)
+			value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -_softclipTable[-value - SOFTCLIP_THRESHOLD];
+
+		_outputSamples.push_back(value);
+	}
+	return frame_size;
+}
+
+int QDM2Stream::readBuffer(int16 *buffer, const int numSamples) {
+	debug(1, "QDM2Stream::readBuffer numSamples: %d", numSamples);
+	int32 decodedSamples = _outputSamples.size();
+	int32 i;
+
+	while (decodedSamples < numSamples) {
+		i = qdm2_decodeFrame(_stream);
+		if (i == 0)
+			break; // Out Of Decode Frames...
+		decodedSamples += i;
+	}
+
+	if (decodedSamples > numSamples)
+		decodedSamples = numSamples;
+
+	for (i = 0; i < decodedSamples; i++)
+		buffer[i] = _outputSamples.remove_at(0);
+
+	return decodedSamples;
+}
+
+AudioStream *makeQDM2Stream(Common::SeekableReadStream *stream, Common::SeekableReadStream *extraData) {
+	return new QDM2Stream(stream, extraData);
+}
+
+} // End of namespace Audio
+
+#endif
diff --git a/audio/decoders/qdm2.h b/audio/decoders/qdm2.h
new file mode 100644
index 0000000000..3d4b542de1
--- /dev/null
+++ b/audio/decoders/qdm2.h
@@ -0,0 +1,52 @@
+/* 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.
+ *
+ * $URL$
+ * $Id$
+ *
+ */
+
+// Only compile if Mohawk is enabled or if we're building dynamic modules
+#if defined(ENABLE_MOHAWK) || defined(DYNAMIC_MODULES)
+
+#ifndef AUDIO_QDM2_H
+#define AUDIO_QDM2_H
+
+namespace Common {
+	class SeekableReadStream;
+}
+
+namespace Audio {
+
+class AudioStream;
+
+/**
+ * Create a new AudioStream from the QDM2 data in the given stream.
+ *
+ * @param stream       the SeekableReadStream from which to read the FLAC data
+ * @param extraData    the QuickTime extra data stream
+ * @return   a new AudioStream, or NULL, if an error occurred
+ */
+AudioStream *makeQDM2Stream(Common::SeekableReadStream *stream, Common::SeekableReadStream *extraData);
+
+} // End of namespace Audio
+
+#endif // AUDIO_QDM2_H
+#endif // Mohawk/Plugins guard
diff --git a/audio/decoders/qdm2data.h b/audio/decoders/qdm2data.h
new file mode 100644
index 0000000000..cfe1a3d10e
--- /dev/null
+++ b/audio/decoders/qdm2data.h
@@ -0,0 +1,531 @@
+/* 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 AUDIO_QDM2DATA_H
+#define AUDIO_QDM2DATA_H
+
+#include "common/scummsys.h"
+
+namespace Audio {
+
+/// VLC TABLES
+
+// values in this table range from -1..23; adjust retrieved value by -1
+static const uint16 vlc_tab_level_huffcodes[24] = {
+	0x037c, 0x0004, 0x003c, 0x004c, 0x003a, 0x002c, 0x001c, 0x001a,
+	0x0024, 0x0014, 0x0001, 0x0002, 0x0000, 0x0003, 0x0007, 0x0005,
+	0x0006, 0x0008, 0x0009, 0x000a, 0x000c, 0x00fc, 0x007c, 0x017c
+};
+
+static const byte vlc_tab_level_huffbits[24] = {
+	10, 6, 7, 7, 6, 6, 6, 6, 6, 5, 4, 4, 4, 3, 3, 3, 3, 4, 4, 5, 7, 8, 9, 10
+};
+
+// values in this table range from -1..36; adjust retrieved value by -1
+static const uint16 vlc_tab_diff_huffcodes[37] = {
+	0x1c57, 0x0004, 0x0000, 0x0001, 0x0003, 0x0002, 0x000f, 0x000e,
+	0x0007, 0x0016, 0x0037, 0x0027, 0x0026, 0x0066, 0x0006, 0x0097,
+	0x0046, 0x01c6, 0x0017, 0x0786, 0x0086, 0x0257, 0x00d7, 0x0357,
+	0x00c6, 0x0386, 0x0186, 0x0000, 0x0157, 0x0c57, 0x0057, 0x0000,
+	0x0b86, 0x0000, 0x1457, 0x0000, 0x0457
+};
+
+static const byte vlc_tab_diff_huffbits[37] = {
+	13, 3, 3, 2, 3, 3, 4, 4, 6, 5, 6, 6, 7, 7, 8, 8,
+	8, 9, 8, 11, 9, 10, 8, 10, 9, 12, 10, 0, 10, 13, 11, 0,
+	12, 0, 13, 0, 13
+};
+
+// values in this table range from -1..5; adjust retrieved value by -1
+static const byte vlc_tab_run_huffcodes[6] = {
+	0x1f, 0x00, 0x01, 0x03, 0x07, 0x0f
+};
+
+static const byte vlc_tab_run_huffbits[6] = {
+	5, 1, 2, 3, 4, 5
+};
+
+// values in this table range from -1..19; adjust retrieved value by -1
+static const uint16 vlc_tab_tone_level_idx_hi1_huffcodes[20] = {
+	0x5714, 0x000c, 0x0002, 0x0001, 0x0000, 0x0004, 0x0034, 0x0054,
+	0x0094, 0x0014, 0x0114, 0x0214, 0x0314, 0x0614, 0x0e14, 0x0f14,
+	0x2714, 0x0714, 0x1714, 0x3714
+};
+
+static const byte vlc_tab_tone_level_idx_hi1_huffbits[20] = {
+	15, 4, 2, 1, 3, 5, 6, 7, 8, 10, 10, 11, 11, 12, 12, 12, 14, 14, 15, 14
+};
+
+// values in this table range from -1..23; adjust retrieved value by -1
+static const uint16 vlc_tab_tone_level_idx_mid_huffcodes[24] = {
+	0x0fea, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000,
+	0x0000, 0x0000, 0x0000, 0x0000, 0x03ea, 0x00ea, 0x002a, 0x001a,
+	0x0006, 0x0001, 0x0000, 0x0002, 0x000a, 0x006a, 0x01ea, 0x07ea
+};
+
+static const byte vlc_tab_tone_level_idx_mid_huffbits[24] = {
+	12, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 11, 9, 7, 5, 3, 1, 2, 4, 6, 8, 10, 12
+};
+
+// values in this table range from -1..23; adjust retrieved value by -1
+static const uint16 vlc_tab_tone_level_idx_hi2_huffcodes[24] = {
+	0x0664, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0064, 0x00e4,
+	0x00a4, 0x0068, 0x0004, 0x0008, 0x0014, 0x0018, 0x0000, 0x0001,
+	0x0002, 0x0003, 0x000c, 0x0028, 0x0024, 0x0164, 0x0000, 0x0264
+};
+
+static const byte vlc_tab_tone_level_idx_hi2_huffbits[24] = {
+	11, 0, 0, 0, 0, 0, 10, 8, 8, 7, 6, 6, 5, 5, 4, 2, 2, 2, 4, 7, 8, 9, 0, 11
+};
+
+// values in this table range from -1..8; adjust retrieved value by -1
+static const byte vlc_tab_type30_huffcodes[9] = {
+	0x3c, 0x06, 0x00, 0x01, 0x03, 0x02, 0x04, 0x0c, 0x1c
+};
+
+static const byte vlc_tab_type30_huffbits[9] = {
+	6, 3, 3, 2, 2, 3, 4, 5, 6
+};
+
+// values in this table range from -1..9; adjust retrieved value by -1
+static const byte vlc_tab_type34_huffcodes[10] = {
+	0x18, 0x00, 0x01, 0x04, 0x05, 0x07, 0x03, 0x02, 0x06, 0x08
+};
+
+static const byte vlc_tab_type34_huffbits[10] = {
+	5, 4, 3, 3, 3, 3, 3, 3, 3, 5
+};
+
+// values in this table range from -1..22; adjust retrieved value by -1
+static const uint16 vlc_tab_fft_tone_offset_0_huffcodes[23] = {
+	0x038e, 0x0001, 0x0000, 0x0022, 0x000a, 0x0006, 0x0012, 0x0002,
+	0x001e, 0x003e, 0x0056, 0x0016, 0x000e, 0x0032, 0x0072, 0x0042,
+	0x008e, 0x004e, 0x00f2, 0x002e, 0x0036, 0x00c2, 0x018e
+};
+
+static const byte vlc_tab_fft_tone_offset_0_huffbits[23] = {
+	10, 1, 2, 6, 4, 5, 6, 7, 6, 6, 7, 7, 8, 7, 8, 8, 9, 7, 8, 6, 6, 8, 10
+};
+
+// values in this table range from -1..27; adjust retrieved value by -1
+static const uint16 vlc_tab_fft_tone_offset_1_huffcodes[28] = {
+	0x07a4, 0x0001, 0x0020, 0x0012, 0x001c, 0x0008, 0x0006, 0x0010,
+	0x0000, 0x0014, 0x0004, 0x0032, 0x0070, 0x000c, 0x0002, 0x003a,
+	0x001a, 0x002c, 0x002a, 0x0022, 0x0024, 0x000a, 0x0064, 0x0030,
+	0x0062, 0x00a4, 0x01a4, 0x03a4
+};
+
+static const byte vlc_tab_fft_tone_offset_1_huffbits[28] = {
+	11, 1, 6, 6, 5, 4, 3, 6, 6, 5, 6, 6, 7, 6, 6, 6,
+	6, 6, 6, 7, 8, 6, 7, 7, 7, 9, 10, 11
+};
+
+// values in this table range from -1..31; adjust retrieved value by -1
+static const uint16 vlc_tab_fft_tone_offset_2_huffcodes[32] = {
+	0x1760, 0x0001, 0x0000, 0x0082, 0x000c, 0x0006, 0x0003, 0x0007,
+	0x0008, 0x0004, 0x0010, 0x0012, 0x0022, 0x001a, 0x0000, 0x0020,
+	0x000a, 0x0040, 0x004a, 0x006a, 0x002a, 0x0042, 0x0002, 0x0060,
+	0x00aa, 0x00e0, 0x00c2, 0x01c2, 0x0160, 0x0360, 0x0760, 0x0f60
+};
+
+static const byte vlc_tab_fft_tone_offset_2_huffbits[32] = {
+	13, 2, 0, 8, 4, 3, 3, 3, 4, 4, 5, 5, 6, 5, 7, 7,
+	7, 7, 7, 7, 8, 8, 8, 9, 8, 8, 9, 9, 10, 11, 13, 12
+};
+
+// values in this table range from -1..34; adjust retrieved value by -1
+static const uint16 vlc_tab_fft_tone_offset_3_huffcodes[35] = {
+	0x33ea, 0x0005, 0x0000, 0x000c, 0x0000, 0x0006, 0x0003, 0x0008,
+	0x0002, 0x0001, 0x0004, 0x0007, 0x001a, 0x000f, 0x001c, 0x002c,
+	0x000a, 0x001d, 0x002d, 0x002a, 0x000d, 0x004c, 0x008c, 0x006a,
+	0x00cd, 0x004d, 0x00ea, 0x020c, 0x030c, 0x010c, 0x01ea, 0x07ea,
+	0x0bea, 0x03ea, 0x13ea
+};
+
+static const byte vlc_tab_fft_tone_offset_3_huffbits[35] = {
+	14, 4, 0, 10, 4, 3, 3, 4, 4, 3, 4, 4, 5, 4, 5, 6,
+	6, 5, 6, 7, 7, 7, 8, 8, 8, 8, 9, 10, 10, 10, 10, 11,
+	12, 13, 14
+};
+
+// values in this table range from -1..37; adjust retrieved value by -1
+static const uint16 vlc_tab_fft_tone_offset_4_huffcodes[38] = {
+	0x5282, 0x0016, 0x0000, 0x0136, 0x0004, 0x0000, 0x0007, 0x000a,
+	0x000e, 0x0003, 0x0001, 0x000d, 0x0006, 0x0009, 0x0012, 0x0005,
+	0x0025, 0x0022, 0x0015, 0x0002, 0x0076, 0x0035, 0x0042, 0x00c2,
+	0x0182, 0x00b6, 0x0036, 0x03c2, 0x0482, 0x01c2, 0x0682, 0x0882,
+	0x0a82, 0x0082, 0x0282, 0x1282, 0x3282, 0x2282
+};
+
+static const byte vlc_tab_fft_tone_offset_4_huffbits[38] = {
+	15, 6, 0, 9, 3, 3, 3, 4, 4, 3, 4, 4, 5, 4, 5, 6,
+	6, 6, 6, 8, 7, 6, 8, 9, 9, 8, 9, 10, 11, 10, 11, 12,
+	12, 12, 14, 15, 14, 14
+};
+
+/// FFT TABLES
+
+// values in this table range from -1..27; adjust retrieved value by -1
+static const uint16 fft_level_exp_alt_huffcodes[28] = {
+	0x1ec6, 0x0006, 0x00c2, 0x0142, 0x0242, 0x0246, 0x00c6, 0x0046,
+	0x0042, 0x0146, 0x00a2, 0x0062, 0x0026, 0x0016, 0x000e, 0x0005,
+	0x0004, 0x0003, 0x0000, 0x0001, 0x000a, 0x0012, 0x0002, 0x0022,
+	0x01c6, 0x02c6, 0x06c6, 0x0ec6
+};
+
+static const byte fft_level_exp_alt_huffbits[28] = {
+	13, 7, 8, 9, 10, 10, 10, 10, 10, 9, 8, 7, 6, 5, 4, 3,
+	3, 2, 3, 3, 4, 5, 7, 8, 9, 11, 12, 13
+};
+
+// values in this table range from -1..19; adjust retrieved value by -1
+static const uint16 fft_level_exp_huffcodes[20] = {
+	0x0f24, 0x0001, 0x0002, 0x0000, 0x0006, 0x0005, 0x0007, 0x000c,
+	0x000b, 0x0014, 0x0013, 0x0004, 0x0003, 0x0023, 0x0064, 0x00a4,
+	0x0024, 0x0124, 0x0324, 0x0724
+};
+
+static const byte fft_level_exp_huffbits[20] = {
+	12, 3, 3, 3, 3, 3, 3, 4, 4, 5, 5, 6, 6, 6, 7, 8, 9, 10, 11, 12
+};
+
+// values in this table range from -1..6; adjust retrieved value by -1
+static const byte fft_stereo_exp_huffcodes[7] = {
+	0x3e, 0x01, 0x00, 0x02, 0x06, 0x0e, 0x1e
+};
+
+static const byte fft_stereo_exp_huffbits[7] = {
+	6, 1, 2, 3, 4, 5, 6
+};
+
+// values in this table range from -1..8; adjust retrieved value by -1
+static const byte fft_stereo_phase_huffcodes[9] = {
+	0x35, 0x02, 0x00, 0x01, 0x0d, 0x15, 0x05, 0x09, 0x03
+};
+
+static const byte fft_stereo_phase_huffbits[9] = {
+	6, 2, 2, 4, 4, 6, 5, 4, 2
+};
+
+static const int fft_cutoff_index_table[4][2] = {
+	{ 1, 2 }, {-1, 0 }, {-1,-2 }, { 0, 0 }
+};
+
+static const int16 fft_level_index_table[256] = {
+	0, 0, 0, 0, 0, 0, 0, 0, 1, 1, 1, 1, 1, 1, 1, 1,
+	2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2,
+	3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3,
+	3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3,
+	4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4,
+	4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4,
+	4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4,
+	4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4,
+	5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
+	5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
+	5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
+	5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
+	5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
+	5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
+	5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
+	5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
+};
+
+static const byte last_coeff[3] = {
+	4, 7, 10
+};
+
+static const byte coeff_per_sb_for_avg[3][30] = {
+	{ 0, 1, 1, 1, 1, 2, 2, 2, 2, 2, 2, 2, 2, 2, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3 },
+	{ 0, 1, 2, 2, 3, 3, 4, 4, 4, 4, 4, 4, 5, 5, 5, 5, 5, 5, 5, 5, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6 },
+	{ 0, 1, 2, 3, 4, 4, 5, 5, 6, 6, 6, 6, 7, 7, 7, 7, 8, 8, 8, 8, 8, 8, 9, 9, 9, 9, 9, 9, 9, 9 }
+};
+
+static const uint32 dequant_table[3][10][30] = {
+	{ { 256, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
+	  { 0, 256, 256, 205, 154, 102, 51, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
+	  { 0, 0, 0, 51, 102, 154, 205, 256, 238, 219, 201, 183, 165, 146, 128, 110, 91, 73, 55, 37, 18, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
+	  { 0, 0, 0, 0, 0, 0, 0, 0, 18, 37, 55, 73, 91, 110, 128, 146, 165, 183, 201, 219, 238, 256, 228, 199, 171, 142, 114, 85, 57, 28 },
+	  { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
+	  { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
+	  { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
+	  { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
+	  { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
+	  { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 } },
+	{ { 256, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
+	  { 0, 256, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
+	  { 0, 0, 256, 171, 85, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
+	  { 0, 0, 0, 85, 171, 256, 171, 85, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
+	  { 0, 0, 0, 0, 0, 0, 85, 171, 256, 219, 183, 146, 110, 73, 37, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
+	  { 0, 0, 0, 0, 0, 0, 0, 0, 0, 37, 73, 110, 146, 183, 219, 256, 228, 199, 171, 142, 114, 85, 57, 28, 0, 0, 0, 0, 0, 0 },
+	  { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 28, 57, 85, 114, 142, 171, 199, 228, 256, 213, 171, 128, 85, 43 },
+	  { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
+	  { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
+	  { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 } },
+	{ { 256, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
+	  { 0, 256, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
+	  { 0, 0, 256, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
+	  { 0, 0, 0, 256, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
+	  { 0, 0, 0, 0, 256, 256, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
+	  { 0, 0, 0, 0, 0, 0, 256, 171, 85, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
+	  { 0, 0, 0, 0, 0, 0, 0, 85, 171, 256, 192, 128, 64, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
+	  { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 64, 128, 192, 256, 205, 154, 102, 51, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
+	  { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 51, 102, 154, 205, 256, 213, 171, 128, 85, 43, 0, 0, 0, 0, 0, 0 },
+	  { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 43, 85, 128, 171, 213, 256, 213, 171, 128, 85, 43 } }
+};
+
+static const byte coeff_per_sb_for_dequant[3][30] = {
+	{ 0, 1, 1, 1, 1, 1, 1, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 3, 3, 3, 3, 3, 3, 3, 3, 3 },
+	{ 0, 1, 2, 2, 2, 3, 3, 3, 4, 4, 4, 4, 4, 4, 4, 5, 5, 5, 5, 5, 5, 5, 5, 5, 6, 6, 6, 6, 6, 6 },
+	{ 0, 1, 2, 3, 4, 4, 5, 5, 5, 6, 6, 6, 6, 7, 7, 7, 7, 7, 8, 8, 8, 8, 8, 8, 9, 9, 9, 9, 9, 9 }
+};
+
+// first index is subband, 2nd index is 0, 1 or 3 (2 is unused)
+static const int8 tone_level_idx_offset_table[30][4] = {
+	{ -50, -50,  0, -50 },
+	{ -50, -50,  0, -50 },
+	{ -50,  -9,  0, -19 },
+	{ -16,  -6,  0, -12 },
+	{ -11,  -4,  0,  -8 },
+	{  -8,  -3,  0,  -6 },
+	{  -7,  -3,  0,  -5 },
+	{  -6,  -2,  0,  -4 },
+	{  -5,  -2,  0,  -3 },
+	{  -4,  -1,  0,  -3 },
+	{  -4,  -1,  0,  -2 },
+	{  -3,  -1,  0,  -2 },
+	{  -3,  -1,  0,  -2 },
+	{  -3,  -1,  0,  -2 },
+	{  -2,  -1,  0,  -1 },
+	{  -2,  -1,  0,  -1 },
+	{  -2,  -1,  0,  -1 },
+	{  -2,   0,  0,  -1 },
+	{  -2,   0,  0,  -1 },
+	{  -1,   0,  0,  -1 },
+	{  -1,   0,  0,  -1 },
+	{  -1,   0,  0,  -1 },
+	{  -1,   0,  0,  -1 },
+	{  -1,   0,  0,  -1 },
+	{  -1,   0,  0,  -1 },
+	{  -1,   0,  0,  -1 },
+	{  -1,   0,  0,   0 },
+	{  -1,   0,  0,   0 },
+	{  -1,   0,  0,   0 },
+	{  -1,   0,  0,   0 }
+};
+
+/* all my samples have 1st index 0 or 1 */
+/* second index is subband, only indexes 0-29 seem to be used */
+static const int8 coding_method_table[5][30] = {
+	{ 34, 30, 24, 24, 16, 16, 16, 16, 10, 10, 10, 10, 10, 10, 10,
+	  10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10
+	},
+	{ 34, 30, 24, 24, 16, 16, 16, 16, 10, 10, 10, 10, 10, 10, 10,
+	  10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10
+	},
+	{ 34, 30, 30, 30, 24, 24, 16, 16, 16, 16, 16, 16, 10, 10, 10,
+	  10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10
+	},
+	{ 34, 34, 30, 30, 24, 24, 24, 24, 16, 16, 16, 16, 16, 16, 16,
+	  16, 16, 16, 16, 16, 16, 16, 10, 10, 10, 10, 10, 10, 10, 10
+	},
+	{ 34, 34, 30, 30, 30, 30, 30, 30, 24, 24, 24, 24, 24, 24, 24,
+	  24, 24, 24, 24, 24, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16
+	},
+};
+
+static const int vlc_stage3_values[60] = {
+	    0,     1,     2,     3,     4,     6,     8,    10,    12,    16,    20,    24,
+	   28,    36,    44,    52,    60,    76,    92,   108,   124,   156,   188,   220,
+	  252,   316,   380,   444,   508,   636,   764,   892,  1020,  1276,  1532,  1788,
+	 2044,  2556,  3068,  3580,  4092,  5116,  6140,  7164,  8188, 10236, 12284, 14332,
+	16380, 20476, 24572, 28668, 32764, 40956, 49148, 57340, 65532, 81916, 98300,114684
+};
+
+static const float fft_tone_sample_table[4][16][5] = {
+	{ { .0100000000f,-.0037037037f,-.0020000000f,-.0069444444f,-.0018416207f },
+	  { .0416666667f, .0000000000f, .0000000000f,-.0208333333f,-.0123456791f },
+	  { .1250000000f, .0558035709f, .0330687836f,-.0164473690f,-.0097465888f },
+	  { .1562500000f, .0625000000f, .0370370370f,-.0062500000f,-.0037037037f },
+	  { .1996007860f, .0781250000f, .0462962948f, .0022727272f, .0013468013f },
+	  { .2000000000f, .0625000000f, .0370370373f, .0208333333f, .0074074073f },
+	  { .2127659619f, .0555555556f, .0329218097f, .0208333333f, .0123456791f },
+	  { .2173913121f, .0473484844f, .0280583613f, .0347222239f, .0205761325f },
+	  { .2173913121f, .0347222239f, .0205761325f, .0473484844f, .0280583613f },
+	  { .2127659619f, .0208333333f, .0123456791f, .0555555556f, .0329218097f },
+	  { .2000000000f, .0208333333f, .0074074073f, .0625000000f, .0370370370f },
+	  { .1996007860f, .0022727272f, .0013468013f, .0781250000f, .0462962948f },
+	  { .1562500000f,-.0062500000f,-.0037037037f, .0625000000f, .0370370370f },
+	  { .1250000000f,-.0164473690f,-.0097465888f, .0558035709f, .0330687836f },
+	  { .0416666667f,-.0208333333f,-.0123456791f, .0000000000f, .0000000000f },
+	  { .0100000000f,-.0069444444f,-.0018416207f,-.0037037037f,-.0020000000f } },
+
+	{ { .0050000000f,-.0200000000f, .0125000000f,-.3030303030f, .0020000000f },
+	  { .1041666642f, .0400000000f,-.0250000000f, .0333333333f,-.0200000000f },
+	  { .1250000000f, .0100000000f, .0142857144f,-.0500000007f,-.0200000000f },
+	  { .1562500000f,-.0006250000f,-.00049382716f,-.000625000f,-.00049382716f },
+	  { .1562500000f,-.0006250000f,-.00049382716f,-.000625000f,-.00049382716f },
+	  { .1250000000f,-.0500000000f,-.0200000000f, .0100000000f, .0142857144f },
+	  { .1041666667f, .0333333333f,-.0200000000f, .0400000000f,-.0250000000f },
+	  { .0050000000f,-.3030303030f, .0020000001f,-.0200000000f, .0125000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f } },
+
+	{ { .1428571492f, .1250000000f,-.0285714287f,-.0357142873f, .0208333333f },
+	  { .1818181818f, .0588235296f, .0333333333f, .0212765951f, .0100000000f },
+	  { .1818181818f, .0212765951f, .0100000000f, .0588235296f, .0333333333f },
+	  { .1428571492f,-.0357142873f, .0208333333f, .1250000000f,-.0285714287f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f } },
+
+	{ { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
+	  { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f } }
+};
+
+static const float fft_tone_level_table[2][64] = { {
+// pow ~ (i > 46) ? 0 : (((((i & 1) ? 431 : 304) << (i >> 1))) / 1024.0);
+	0.17677669f, 0.42677650f, 0.60355347f, 0.85355347f,
+	1.20710683f, 1.68359375f, 2.37500000f, 3.36718750f,
+	4.75000000f, 6.73437500f, 9.50000000f, 13.4687500f,
+	19.0000000f, 26.9375000f, 38.0000000f, 53.8750000f,
+	76.0000000f, 107.750000f, 152.000000f, 215.500000f,
+	304.000000f, 431.000000f, 608.000000f, 862.000000f,
+	1216.00000f, 1724.00000f, 2432.00000f, 3448.00000f,
+	4864.00000f, 6896.00000f, 9728.00000f, 13792.0000f,
+	19456.0000f, 27584.0000f, 38912.0000f, 55168.0000f,
+	77824.0000f, 110336.000f, 155648.000f, 220672.000f,
+	311296.000f, 441344.000f, 622592.000f, 882688.000f,
+	1245184.00f, 1765376.00f, 2490368.00f, 0.00000000f,
+	0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
+	0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
+	0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
+	0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
+  }, {
+// pow = (i > 45) ? 0 : ((((i & 1) ? 431 : 304) << (i >> 1)) / 512.0);
+	0.59375000f, 0.84179688f, 1.18750000f, 1.68359375f,
+	2.37500000f, 3.36718750f, 4.75000000f, 6.73437500f,
+	9.50000000f, 13.4687500f, 19.0000000f, 26.9375000f,
+	38.0000000f, 53.8750000f, 76.0000000f, 107.750000f,
+	152.000000f, 215.500000f, 304.000000f, 431.000000f,
+	608.000000f, 862.000000f, 1216.00000f, 1724.00000f,
+	2432.00000f, 3448.00000f, 4864.00000f, 6896.00000f,
+	9728.00000f, 13792.0000f, 19456.0000f, 27584.0000f,
+	38912.0000f, 55168.0000f, 77824.0000f, 110336.000f,
+	155648.000f, 220672.000f, 311296.000f, 441344.000f,
+	622592.000f, 882688.000f, 1245184.00f, 1765376.00f,
+	2490368.00f, 3530752.00f, 0.00000000f, 0.00000000f,
+	0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
+	0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
+	0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
+	0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f
+} };
+
+static const float fft_tone_envelope_table[4][31] = {
+	{ .009607375f, .038060248f, .084265202f, .146446645f, .222214907f, .308658302f,
+	  .402454883f, .500000060f, .597545207f, .691341758f, .777785182f, .853553414f,
+	  .915734828f, .961939812f, .990392685f, 1.00000000f, .990392625f, .961939752f,
+	  .915734768f, .853553295f, .777785063f, .691341639f, .597545087f, .500000000f,
+	  .402454853f, .308658272f, .222214878f, .146446615f, .084265172f, .038060218f,
+	  .009607345f },
+	{ .038060248f, .146446645f, .308658302f, .500000060f, .691341758f, .853553414f,
+	  .961939812f, 1.00000000f, .961939752f, .853553295f, .691341639f, .500000000f,
+	  .308658272f, .146446615f, .038060218f, .000000000f, .000000000f, .000000000f,
+	  .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f,
+	  .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f,
+	  .000000000f },
+	{ .146446645f, .500000060f, .853553414f, 1.00000000f, .853553295f, .500000000f,
+	  .146446615f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f,
+	  .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f,
+	  .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f,
+	  .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f,
+	  .000000000f },
+	{ .500000060f, 1.00000000f, .500000000f, .000000000f, .000000000f, .000000000f,
+	  .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f,
+	  .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f,
+	  .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f,
+	  .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f,
+	  .000000000f }
+};
+
+static const float sb_noise_attenuation[32] = {
+	0.0f, 0.0f, 0.3f, 0.4f, 0.5f, 0.7f, 1.0f, 1.0f,
+	1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f,
+	1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f,
+	1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f,
+};
+
+static const byte fft_subpackets[32] = {
+	0, 0, 0, 0, 0, 0, 0, 0, 1, 1, 1, 1, 1, 1, 1, 0,
+	0, 0, 0, 0, 0, 0, 0, 0, 1, 1, 1, 1, 1, 1, 0, 0
+};
+
+// first index is joined_stereo, second index is 0 or 2 (1 is unused)
+static const float dequant_1bit[2][3] = {
+	{-0.920000f, 0.000000f, 0.920000f },
+	{-0.890000f, 0.000000f, 0.890000f }
+};
+
+static const float type30_dequant[8] = {
+	-1.0f,-0.625f,-0.291666656732559f,0.0f,
+	0.25f,0.5f,0.75f,1.0f,
+};
+
+static const float type34_delta[10] = { // FIXME: covers 8 entries..
+	-1.0f,-0.60947573184967f,-0.333333343267441f,-0.138071194291115f,0.0f,
+	0.138071194291115f,0.333333343267441f,0.60947573184967f,1.0f,0.0f,
+};
+
+} // End of namespace Audio
+
+#endif
diff --git a/audio/decoders/quicktime.cpp b/audio/decoders/quicktime.cpp
new file mode 100644
index 0000000000..af59d8803f
--- /dev/null
+++ b/audio/decoders/quicktime.cpp
@@ -0,0 +1,311 @@
+/* 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/memstream.h"
+#include "common/stream.h"
+
+#include "audio/audiostream.h"
+#include "audio/decoders/quicktime.h"
+
+// Codecs
+#include "audio/decoders/adpcm.h"
+#include "audio/decoders/qdm2.h"
+#include "audio/decoders/raw.h"
+
+namespace Audio {
+
+QuickTimeAudioDecoder::QuickTimeAudioDecoder() : Common::QuickTimeParser() {
+	_audStream = 0;
+}
+
+QuickTimeAudioDecoder::~QuickTimeAudioDecoder() {
+}
+
+bool QuickTimeAudioDecoder::loadFile(const Common::String &filename) {
+	if (!Common::QuickTimeParser::loadFile(filename))
+		return false;
+
+	init();
+	return true;
+}
+
+bool QuickTimeAudioDecoder::loadStream(Common::SeekableReadStream *stream) {
+	if (!Common::QuickTimeParser::loadStream(stream))
+		return false;
+
+	init();
+	return true;
+}
+
+void QuickTimeAudioDecoder::init() {
+	Common::QuickTimeParser::init();
+
+	_audioStreamIndex = -1;
+
+	// Find an audio stream
+	for (uint32 i = 0; i < _numStreams; i++)
+		if (_streams[i]->codec_type == CODEC_TYPE_AUDIO && _audioStreamIndex < 0)
+			_audioStreamIndex = i;
+
+	// Initialize audio, if present
+	if (_audioStreamIndex >= 0) {
+		AudioSampleDesc *entry = (AudioSampleDesc *)_streams[_audioStreamIndex]->sampleDescs[0];
+
+		if (checkAudioCodecSupport(entry->codecTag)) {
+			_audStream = makeQueuingAudioStream(entry->sampleRate, entry->channels == 2);
+			_curAudioChunk = 0;
+
+			// Make sure the bits per sample transfers to the sample size
+			if (entry->codecTag == MKID_BE('raw ') || entry->codecTag == MKID_BE('twos'))
+				_streams[_audioStreamIndex]->sample_size = (entry->bitsPerSample / 8) * entry->channels;
+		}
+	}
+}
+
+Common::QuickTimeParser::SampleDesc *QuickTimeAudioDecoder::readSampleDesc(MOVStreamContext *st, uint32 format) {
+	if (st->codec_type == CODEC_TYPE_AUDIO) {
+		debug(0, "Audio Codec FourCC: \'%s\'", tag2str(format));
+
+		AudioSampleDesc *entry = new AudioSampleDesc();
+		entry->codecTag = format;
+
+		uint16 stsdVersion = _fd->readUint16BE();
+		_fd->readUint16BE(); // revision level
+		_fd->readUint32BE(); // vendor
+
+		entry->channels = _fd->readUint16BE();			 // channel count
+		entry->bitsPerSample = _fd->readUint16BE();	  // sample size
+
+		_fd->readUint16BE(); // compression id = 0
+		_fd->readUint16BE(); // packet size = 0
+
+		entry->sampleRate = (_fd->readUint32BE() >> 16);
+
+		debug(0, "stsd version =%d", stsdVersion);
+		if (stsdVersion == 0) {
+			// Not used, except in special cases. See below.
+			entry->samplesPerFrame = entry->bytesPerFrame = 0;
+		} else if (stsdVersion == 1) {
+			// Read QT version 1 fields. In version 0 these dont exist.
+			entry->samplesPerFrame = _fd->readUint32BE();
+			debug(0, "stsd samples_per_frame =%d",entry->samplesPerFrame);
+			_fd->readUint32BE(); // bytes per packet
+			entry->bytesPerFrame = _fd->readUint32BE();
+			debug(0, "stsd bytes_per_frame =%d", entry->bytesPerFrame);
+			_fd->readUint32BE(); // bytes per sample
+		} else {
+			warning("Unsupported QuickTime STSD audio version %d", stsdVersion);
+			delete entry;
+			return 0;
+		}
+
+		// Version 0 videos (such as the Riven ones) don't have this set,
+		// but we need it later on. Add it in here.
+		if (format == MKID_BE('ima4')) {
+			entry->samplesPerFrame = 64;
+			entry->bytesPerFrame = 34 * entry->channels;
+		}
+
+		if (entry->sampleRate == 0 && st->time_scale > 1)
+			entry->sampleRate = st->time_scale;
+
+		return entry;
+	}
+
+	return 0;
+}
+
+bool QuickTimeAudioDecoder::checkAudioCodecSupport(uint32 tag) {
+	// Check if the codec is a supported codec
+	if (tag == MKID_BE('twos') || tag == MKID_BE('raw ') || tag == MKID_BE('ima4'))
+		return true;
+
+#ifdef AUDIO_QDM2_H
+	if (tag == MKID_BE('QDM2'))
+		return true;
+#endif
+
+	if (tag == MKID_BE('mp4a'))
+		warning("No MPEG-4 audio (AAC) support");
+	else
+		warning("Audio Codec Not Supported: \'%s\'", tag2str(tag));
+
+	return false;
+}
+
+AudioStream *QuickTimeAudioDecoder::createAudioStream(Common::SeekableReadStream *stream) {
+	if (!stream || _audioStreamIndex < 0)
+		return NULL;
+
+	AudioSampleDesc *entry = (AudioSampleDesc *)_streams[_audioStreamIndex]->sampleDescs[0];
+
+	if (entry->codecTag == MKID_BE('twos') || entry->codecTag == MKID_BE('raw ')) {
+		// Fortunately, most of the audio used in Myst videos is raw...
+		uint16 flags = 0;
+		if (entry->codecTag == MKID_BE('raw '))
+			flags |= FLAG_UNSIGNED;
+		if (entry->channels == 2)
+			flags |= FLAG_STEREO;
+		if (entry->bitsPerSample == 16)
+			flags |= FLAG_16BITS;
+		uint32 dataSize = stream->size();
+		byte *data = (byte *)malloc(dataSize);
+		stream->read(data, dataSize);
+		delete stream;
+		return makeRawStream(data, dataSize, entry->sampleRate, flags);
+	} else if (entry->codecTag == MKID_BE('ima4')) {
+		// Riven uses this codec (as do some Myst ME videos)
+		return makeADPCMStream(stream, DisposeAfterUse::YES, stream->size(), kADPCMApple, entry->sampleRate, entry->channels, 34);
+#ifdef AUDIO_QDM2_H
+	} else if (entry->codecTag == MKID_BE('QDM2')) {
+		// Myst ME uses this codec for many videos
+		return makeQDM2Stream(stream, _streams[_audioStreamIndex]->extradata);
+#endif
+	}
+
+	error("Unsupported audio codec");
+
+	return NULL;
+}
+
+uint32 QuickTimeAudioDecoder::getAudioChunkSampleCount(uint chunk) {
+	if (_audioStreamIndex < 0)
+		return 0;
+
+	uint32 sampleCount = 0;
+
+	for (uint32 j = 0; j < _streams[_audioStreamIndex]->sample_to_chunk_sz; j++)
+		if (chunk >= _streams[_audioStreamIndex]->sample_to_chunk[j].first)
+			sampleCount = _streams[_audioStreamIndex]->sample_to_chunk[j].count;
+
+	return sampleCount;
+}
+
+void QuickTimeAudioDecoder::queueNextAudioChunk() {
+	AudioSampleDesc *entry = (AudioSampleDesc *)_streams[_audioStreamIndex]->sampleDescs[0];
+	Common::MemoryWriteStreamDynamic *wStream = new Common::MemoryWriteStreamDynamic();
+
+	_fd->seek(_streams[_audioStreamIndex]->chunk_offsets[_curAudioChunk]);
+
+	// First, we have to get the sample count
+	uint32 sampleCount = getAudioChunkSampleCount(_curAudioChunk);
+	assert(sampleCount);
+
+	// Then calculate the right sizes
+	while (sampleCount > 0) {
+		uint32 samples = 0, size = 0;
+
+		if (entry->samplesPerFrame >= 160) {
+			samples = entry->samplesPerFrame;
+			size = entry->bytesPerFrame;
+		} else if (entry->samplesPerFrame > 1) {
+			samples = MIN<uint32>((1024 / entry->samplesPerFrame) * entry->samplesPerFrame, sampleCount);
+			size = (samples / entry->samplesPerFrame) * entry->bytesPerFrame;
+		} else {
+			samples = MIN<uint32>(1024, sampleCount);
+			size = samples * _streams[_audioStreamIndex]->sample_size;
+		}
+
+		// Now, we read in the data for this data and output it
+		byte *data = (byte *)malloc(size);
+		_fd->read(data, size);
+		wStream->write(data, size);
+		free(data);
+		sampleCount -= samples;
+	}
+
+	// Now queue the buffer
+	_audStream->queueAudioStream(createAudioStream(new Common::MemoryReadStream(wStream->getData(), wStream->size(), DisposeAfterUse::YES)));
+	delete wStream;
+
+	_curAudioChunk++;
+}
+
+QuickTimeAudioDecoder::AudioSampleDesc::AudioSampleDesc() : Common::QuickTimeParser::SampleDesc() {
+	channels = 0;
+	sampleRate = 0;
+	samplesPerFrame = 0;
+	bytesPerFrame = 0;
+}
+
+/**
+ * A wrapper around QuickTimeAudioDecoder that implements the RewindableAudioStream API
+ */
+class QuickTimeAudioStream : public RewindableAudioStream, public QuickTimeAudioDecoder {
+public:
+	QuickTimeAudioStream() {}
+
+	~QuickTimeAudioStream() {
+		delete _audStream;
+	}
+
+	bool loadFile(const Common::String &filename) {
+		return QuickTimeAudioDecoder::loadFile(filename) && _audioStreamIndex >= 0 && _audStream;
+	}
+
+	// AudioStream API
+	int readBuffer(int16 *buffer, const int numSamples) {
+		int samples = 0;
+
+		while (samples < numSamples && !endOfData()) {
+			if (_audStream->numQueuedStreams() == 0)
+				queueNextAudioChunk();
+
+			samples += _audStream->readBuffer(buffer + samples, numSamples - samples);
+		}
+
+		return samples;
+	}
+
+	bool isStereo() const { return _audStream->isStereo(); }
+	int getRate() const { return _audStream->getRate(); }
+	bool endOfData() const { return _curAudioChunk >= _streams[_audioStreamIndex]->chunk_count && _audStream->endOfData(); }
+
+	// RewindableAudioStream API
+	bool rewind() {
+		// Reset our parent stream
+		_curAudioChunk = 0;
+		delete _audStream;
+
+		AudioSampleDesc *entry = (AudioSampleDesc *)_streams[_audioStreamIndex]->sampleDescs[0];
+		_audStream = makeQueuingAudioStream(entry->sampleRate, entry->channels == 2);
+		return true;
+	}
+};
+
+RewindableAudioStream *makeQuickTimeStream(const Common::String &filename) {
+	QuickTimeAudioStream *audioStream = new QuickTimeAudioStream();
+
+	if (!audioStream->loadFile(filename)) {
+		delete audioStream;
+		return 0;
+	}
+
+	return audioStream;
+}
+
+} // End of namespace Audio
diff --git a/audio/decoders/quicktime.h b/audio/decoders/quicktime.h
new file mode 100644
index 0000000000..eb0f791748
--- /dev/null
+++ b/audio/decoders/quicktime.h
@@ -0,0 +1,96 @@
+/* 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 AUDIO_QUICKTIME_H
+#define AUDIO_QUICKTIME_H
+
+#include "common/quicktime.h"
+#include "common/scummsys.h"
+#include "common/types.h"
+
+namespace Common {
+	class SeekableReadStream;
+	class String;
+}
+
+namespace Audio {
+
+class AudioStream;
+class RewindableAudioStream;
+class QueuingAudioStream;
+
+class QuickTimeAudioDecoder : public Common::QuickTimeParser {
+public:
+	QuickTimeAudioDecoder();
+	virtual ~QuickTimeAudioDecoder();
+
+	/**
+	 * Load a QuickTime audio file
+	 * @param filename	the filename to load
+	 */
+	bool loadFile(const Common::String &filename);
+
+	/**
+	 * Load a QuickTime audio file from a SeekableReadStream
+	 * @param stream	the stream to load
+	 */
+	bool loadStream(Common::SeekableReadStream *stream);
+
+protected:
+	struct AudioSampleDesc : public Common::QuickTimeParser::SampleDesc {
+		AudioSampleDesc();
+
+		uint16 channels;
+		uint32 sampleRate;
+		uint32 samplesPerFrame;
+		uint32 bytesPerFrame;
+	};
+
+	// Common::QuickTimeParser API
+	virtual Common::QuickTimeParser::SampleDesc *readSampleDesc(MOVStreamContext *st, uint32 format);
+
+	AudioStream *createAudioStream(Common::SeekableReadStream *stream);
+	bool checkAudioCodecSupport(uint32 tag);
+	void init();
+
+	void queueNextAudioChunk();
+	uint32 getAudioChunkSampleCount(uint chunk);
+	int8 _audioStreamIndex;
+	uint _curAudioChunk;
+	QueuingAudioStream *_audStream;
+};
+	
+/**
+ * Try to load a QuickTime sound file from the given file name and create a RewindableAudioStream
+ * from that data.
+ *
+ * @param filename			the filename of the file from which to read the data
+ * @return	a new RewindableAudioStream, or NULL, if an error occurred
+ */
+RewindableAudioStream *makeQuickTimeStream(const Common::String &filename);
+
+} // End of namespace Audio
+
+#endif
diff --git a/audio/module.mk b/audio/module.mk
index a9d9bfc869..a13eb49bb1 100644
--- a/audio/module.mk
+++ b/audio/module.mk
@@ -19,6 +19,8 @@ MODULE_OBJS := \
 	decoders/iff_sound.o \
 	decoders/mac_snd.o \
 	decoders/mp3.o \
+	decoders/qdm2.o \
+	decoders/quicktime.o \
 	decoders/raw.o \
 	decoders/vag.o \
 	decoders/voc.o \