[Scummvm-cvs-logs] SF.net SVN: scummvm:[49171] scummvm/trunk
mthreepwood at users.sourceforge.net
mthreepwood at users.sourceforge.net
Sun May 23 23:41:13 CEST 2010
Revision: 49171
http://scummvm.svn.sourceforge.net/scummvm/?rev=49171&view=rev
Author: mthreepwood
Date: 2010-05-23 21:41:13 +0000 (Sun, 23 May 2010)
Log Message:
-----------
Move the QDM2 code to the graphics module, removing the cyclic dependency.
Modified Paths:
--------------
scummvm/trunk/graphics/module.mk
scummvm/trunk/graphics/video/qt_decoder.cpp
scummvm/trunk/sound/module.mk
Added Paths:
-----------
scummvm/trunk/graphics/video/codecs/qdm2.cpp
scummvm/trunk/graphics/video/codecs/qdm2.h
scummvm/trunk/graphics/video/codecs/qdm2data.h
Removed Paths:
-------------
scummvm/trunk/sound/decoders/qdm2.cpp
scummvm/trunk/sound/decoders/qdm2.h
scummvm/trunk/sound/decoders/qdm2data.h
Modified: scummvm/trunk/graphics/module.mk
===================================================================
--- scummvm/trunk/graphics/module.mk 2010-05-23 19:54:17 UTC (rev 49170)
+++ scummvm/trunk/graphics/module.mk 2010-05-23 21:41:13 UTC (rev 49171)
@@ -32,6 +32,7 @@
video/codecs/mjpeg.o \
video/codecs/msrle.o \
video/codecs/msvideo1.o \
+ video/codecs/qdm2.o \
video/codecs/qtrle.o \
video/codecs/rpza.o \
video/codecs/smc.o \
Copied: scummvm/trunk/graphics/video/codecs/qdm2.cpp (from rev 49170, scummvm/trunk/sound/decoders/qdm2.cpp)
===================================================================
--- scummvm/trunk/graphics/video/codecs/qdm2.cpp (rev 0)
+++ scummvm/trunk/graphics/video/codecs/qdm2.cpp 2010-05-23 21:41:13 UTC (rev 49171)
@@ -0,0 +1,3327 @@
+/* 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 "graphics/video/codecs/qdm2.h"
+
+#ifdef GRAPHICS_QDM2_H
+
+#include "sound/audiostream.h"
+#include "graphics/video/codecs/qdm2data.h"
+
+#include "common/array.h"
+#include "common/stream.h"
+#include "common/system.h"
+
+namespace Graphics {
+
+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 Audio::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)); }
+ 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
+ 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);
+
+ RDFTContext _rdftCtx;
+
+ 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;
+
+ debug(1, "void initGetBits(GetBitContext *s, const uint8 *buffer, int bitSize)");
+
+ 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) {
+ debug(1, "int getBitsCount(GetBitContext *s)");
+ return s->index;
+}
+
+static inline unsigned int getBits1(GetBitContext *s) {
+ int index;
+ uint8 result;
+
+ debug(1, "unsigned int getBits1(GetBitContext *s)");
+
+ index = s->index;
+ result = s->buffer[index >> 3];
+
+ debug(1, "index : %d", index);
+
+ result >>= (index & 0x07);
+ result &= 1;
+ index++;
+ s->index = index;
+
+ return result;
+}
+
+static inline unsigned int getBits(GetBitContext *s, int n) {
+ int tmp, reCache, reIndex;
+
+ debug(1, "unsigned int getBits(GetBitContext *s, int n)");
+
+ reIndex = s->index;
+
+ debug(1, "reIndex : %d", reIndex);
+
+ 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) {
+ int reIndex, reCache;
+
+ debug(1, "void skipBits(GetBitContext *s, int n)");
+
+ reIndex = s->index;
+ reCache = 0;
+
+ debug(1, "reIndex : %d", reIndex);
+
+ reCache = READ_LE_UINT32((const uint8 *)s->buffer + (reIndex >> 3)) >> (reIndex & 0x07);
+ s->index = reIndex + 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 * 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 * 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 * 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;
+
+ debug(1, "int getVlc2(GetBitContext *s, int16 (*table)[2], int bits, int maxDepth)");
+
+ 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];
+
+ debug(1, "reIndex : %d", reIndex);
+ debug(1, "reCache : %d", reCache);
+ debug(1, "index : %d", index);
+ debug(1, "code : %d", code);
+ debug(1, "n : %d", n);
+
+ 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;
+ 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);
+ vlc->table = (int16 (*)[2])realloc(vlc->table, sizeof(int16 *) * 2 * vlc->table_allocated);
+ if (!vlc->table)
+ return -1;
+ }
+ 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);
+ debug(2, "QDM2 new table index=%d size=%d code_prefix=%x n=%d", table_index, table_size, code_prefix, n_prefix);
+ 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);
+ debug(2, "QDM2 i=%d n=%d code=0x%x", i, n, code);
+ // 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);
+ debug(2, "QDM2 %4x: code=%d n=%d",j, i, 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);
+ debug(2, "QDM2 %4x: n=%d (subtable)", j, n);
+ // 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");
+ }
+
+ debug(2, "QDM2 build table nb_codes=%d", nb_codes);
+
+ 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;
+ 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
+ }
+
+ debug(1, "QDM2 Subpacket: type=%d size=%d start_offs=%x", sub_packet->type, sub_packet->size, getBitsCount(gb) / 8);
+}
+
+/**
+ * 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) {
+ debug(1, "QDM2Stream::qdm2_fft_decode_tones() duration: %d b:%d", duration, 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) {
+ debug(1, "QDM2Stream::qdm2_fft_decode_tones() local_int_8: %d", local_int_8);
+ while ((n = qdm2_get_vlc(gb, &_vlcTabFftToneOffset[local_int_8], 1, 2)) < 2) {
+ debug(1, "QDM2Stream::qdm2_fft_decode_tones() local_int_8: %d", local_int_8);
+ 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) {
+ debug(1, "QDM2Stream::qdm2_decode_fft_packets()");
+ 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
+ debug(1, "QDM2Stream::qdm2_decode_fft_packets initGetBits() packet->size*8: %d", packet->size*8);
+ 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?
+ debug(1, "QDM2Stream::qdm2_decode_fft_packets qdm2_fft_decode_tones() #1");
+ qdm2_fft_decode_tones(duration, &gb, unknown_flag);
+ }
+ } else if (type == 31) {
+ for (j=0; j < 4; j++) {
+ debug(1, "QDM2Stream::qdm2_decode_fft_packets qdm2_fft_decode_tones() #2");
+ 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++) {
+ debug(1, "QDM2Stream::qdm2_decode_fft_packets qdm2_fft_decode_tones() #3");
+ 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 * 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 * 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) {
+ debug(1, "QDM2Stream::qdm2_calculate_fft channel: %d", channel);
+ const float gain = (_channels == 1 && _channels == 2) ? 0.5f : 1.0f;
+ int i;
+
+ _fft.complex[channel][0].re *= 2.0f;
+ _fft.complex[channel][0].im = 0.0f;
+
+ //debug(1, "QDM2Stream::qdm2_calculate_fft _fft.complex[channel][0].re: %lf", _fft.complex[channel][0].re);
+ //debug(1, "QDM2Stream::qdm2_calculate_fft _fft.complex[channel][0].im: %lf", _fft.complex[channel][0].im);
+
+ 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] * gain;
+}
+
+/**
+ * @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);
+
+ // select input buffer
+ if(in->eos() || in->size() == in->pos()) {
+ 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;
+ }
+
+ 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");
+
+ // 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");
+
+/*
+ debugN(1, "Input Data Packet:");
+ for(i = 0; i < _packetSize; i++) {
+ debugN(1, " %d", _compressedData[i]);
+ }
+ debugN(1, " Output Data Packet:");
+ for(i = 0; i < frame_size; i++) {
+ debugN(1, " %d", (int)_outputBuffer[i]);
+ }
+ debug(1, "");
+*/
+
+ for (i = 0; i < frame_size; i++) {
+ //debug(1, "QDM2Stream::qdm2_decodeFrame i: %d", 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((int)_outputSamples.size() < numSamples) {
+ while(!_stream->eos() && _stream->pos() != _stream->size()) {
+ 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;
+}
+
+Audio::AudioStream *makeQDM2Stream(Common::SeekableReadStream *stream, Common::SeekableReadStream *extraData) {
+ return new QDM2Stream(stream, extraData);
+}
+
+} // End of namespace Graphics
+
+#endif
Copied: scummvm/trunk/graphics/video/codecs/qdm2.h (from rev 49170, scummvm/trunk/sound/decoders/qdm2.h)
===================================================================
--- scummvm/trunk/graphics/video/codecs/qdm2.h (rev 0)
+++ scummvm/trunk/graphics/video/codecs/qdm2.h 2010-05-23 21:41:13 UTC (rev 49171)
@@ -0,0 +1,54 @@
+/* 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 GRAPHICS_QDM2_H
+#define GRAPHICS_QDM2_H
+
+namespace Common {
+ class SeekableReadStream;
+}
+
+namespace Audio {
+ class AudioStream;
+}
+
+namespace Graphics {
+
+/**
+ * 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 occured
+ */
+Audio::AudioStream *makeQDM2Stream(Common::SeekableReadStream *stream, Common::SeekableReadStream *extraData);
+
+} // End of namespace Graphics
+
+#endif // GRAPHICS_QDM2_H
+#endif // Mohawk/Plugins guard
Copied: scummvm/trunk/graphics/video/codecs/qdm2data.h (from rev 49170, scummvm/trunk/sound/decoders/qdm2data.h)
===================================================================
--- scummvm/trunk/graphics/video/codecs/qdm2data.h (rev 0)
+++ scummvm/trunk/graphics/video/codecs/qdm2data.h 2010-05-23 21:41:13 UTC (rev 49171)
@@ -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 GRAPHICS_QDM2DATA_H
+#define GRAPHICS_QDM2DATA_H
+
+#include "common/scummsys.h"
+
+namespace Graphics {
+
+/// 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,
@@ Diff output truncated at 100000 characters. @@
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