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scummvm/audio/decoders/qdm2.cpp

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/* 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.
*
*/
// Based off ffmpeg's QDM2 decoder
#include "common/scummsys.h"
#include "audio/decoders/qdm2.h"
#ifdef AUDIO_QDM2_H
#include "audio/audiostream.h"
#include "audio/decoders/codec.h"
#include "audio/decoders/qdm2data.h"
#include "audio/decoders/raw.h"
#include "common/array.h"
#include "common/debug.h"
#include "common/math.h"
#include "common/rdft.h"
#include "common/stream.h"
#include "common/memstream.h"
#include "common/bitstream.h"
#include "common/textconsole.h"
namespace Audio {
enum {
SOFTCLIP_THRESHOLD = 27600,
HARDCLIP_THRESHOLD = 35716,
MPA_MAX_CHANNELS = 2,
MPA_FRAME_SIZE = 1152,
FF_INPUT_BUFFER_PADDING_SIZE = 8
};
typedef int8 sb_int8_array[2][30][64];
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"
class QDM2Stream : public Codec {
public:
QDM2Stream(Common::SeekableReadStream *extraData, DisposeAfterUse::Flag disposeExtraData);
~QDM2Stream();
AudioStream *decodeFrame(Common::SeekableReadStream &stream);
private:
// 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];
Common::RDFT *_rdft;
QDM2FFT _fft;
// I/O data
uint8 *_compressedData;
float _outputBuffer[1024];
// Synthesis filter
int16 ff_mpa_synth_window[512];
int16 _synthBuf[MPA_MAX_CHANNELS][512*2];
int _synthBufOffset[MPA_MAX_CHANNELS];
int32 _sbSamples[MPA_MAX_CHANNELS][128][32];
// Mixed temporary data used in decoding
float _toneLevel[MPA_MAX_CHANNELS][30][64];
int8 _codingMethod[MPA_MAX_CHANNELS][30][64];
int8 _quantizedCoeffs[MPA_MAX_CHANNELS][10][8];
int8 _toneLevelIdxBase[MPA_MAX_CHANNELS][30][8];
int8 _toneLevelIdxHi1[MPA_MAX_CHANNELS][3][8][8];
int8 _toneLevelIdxMid[MPA_MAX_CHANNELS][26][8];
int8 _toneLevelIdxHi2[MPA_MAX_CHANNELS][26];
int8 _toneLevelIdx[MPA_MAX_CHANNELS][30][64];
int8 _toneLevelIdxTemp[MPA_MAX_CHANNELS][30][64];
// Flags
bool _hasErrors; // packet has errors
int _superblocktype_2_3; // select fft tables and some algorithm based on superblock type
int _doSynthFilter; // used to perform or skip synthesis filter
uint8 _subPacket; // 0 to 15
uint32 _superBlockStart;
int _noiseIdx; // index for dithering noise table
byte _emptyBuffer[FF_INPUT_BUFFER_PADDING_SIZE];
VLC _vlcTabLevel;
VLC _vlcTabDiff;
VLC _vlcTabRun;
VLC _fftLevelExpAltVlc;
VLC _fftLevelExpVlc;
VLC _fftStereoExpVlc;
VLC _fftStereoPhaseVlc;
VLC _vlcTabToneLevelIdxHi1;
VLC _vlcTabToneLevelIdxMid;
VLC _vlcTabToneLevelIdxHi2;
VLC _vlcTabType30;
VLC _vlcTabType34;
VLC _vlcTabFftToneOffset[5];
bool _vlcsInitialized;
void initVlc(void);
uint16 _softclipTable[HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1];
void softclipTableInit(void);
float _noiseTable[4096];
byte _randomDequantIndex[256][5];
byte _randomDequantType24[128][3];
void rndTableInit(void);
float _noiseSamples[128];
void initNoiseSamples(void);
void average_quantized_coeffs(void);
void build_sb_samples_from_noise(int sb);
void fix_coding_method_array(int sb, int channels, sb_int8_array coding_method);
void fill_tone_level_array(int flag);
void fill_coding_method_array(sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
sb_int8_array coding_method, int nb_channels,
int c, int superblocktype_2_3, int cm_table_select);
void synthfilt_build_sb_samples(Common::BitStream *gb, int length, int sb_min, int sb_max);
void init_quantized_coeffs_elem0(int8 *quantized_coeffs, Common::BitStream *gb, int length);
void init_tone_level_dequantization(Common::BitStream *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, Common::BitStream *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);
bool qdm2_decodeFrame(Common::SeekableReadStream &in, QueuingAudioStream *audioStream);
};
// Fix compilation for non C99-compliant compilers, like MSVC
#ifndef int64_t
typedef signed long long int int64_t;
#endif
#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)])
// 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(Common::BitStream *s, int16 (*table)[2], int bits, int maxDepth) {
int index = s->peekBits(bits);
int code = table[index][0];
int n = table[index][1];
if (maxDepth > 1 && n < 0) {
s->skip(bits);
int nbBits = -n;
index = s->peekBits(-n) + code;
code = table[index][0];
n = table[index][1];
if (maxDepth > 2 && n < 0) {
s->skip(nbBits);
index = s->getBits(-n) + code;
code = table[index][0];
n = table[index][1];
}
}
s->skip(n);
return code;
}
static int allocTable(VLC *vlc, int size, int use_static) {
int index;
int16 (*temp)[2] = NULL;
index = vlc->table_size;
vlc->table_size += size;
if (vlc->table_size > vlc->table_allocated) {
if(use_static)
error("QDM2 cant do anything, init_vlc() is used with too little memory");
vlc->table_allocated += (1 << vlc->bits);
temp = (int16 (*)[2])realloc(vlc->table, sizeof(int16 *) * 2 * vlc->table_allocated);
if (!temp) {
free(vlc->table);
vlc->table = NULL;
return -1;
}
vlc->table = temp;
}
return index;
}
#define GET_DATA(v, table, i, wrap, size)\
{\
const uint8 *ptr = (const uint8 *)table + i * wrap;\
switch(size) {\
case 1:\
v = *(const uint8 *)ptr;\
break;\
case 2:\
v = *(const uint16 *)ptr;\
break;\
default:\
v = *(const uint32 *)ptr;\
break;\
}\
}
static int build_table(VLC *vlc, int table_nb_bits,
int nb_codes,
const void *bits, int bits_wrap, int bits_size,
const void *codes, int codes_wrap, int codes_size,
const void *symbols, int symbols_wrap, int symbols_size,
int code_prefix, int n_prefix, int flags)
{
int i, j, k, n, table_size, table_index, nb, n1, index, code_prefix2, symbol;
uint32 code;
int16 (*table)[2];
table_size = 1 << table_nb_bits;
table_index = allocTable(vlc, table_size, flags & 4);
if (table_index < 0)
return -1;
table = &vlc->table[table_index];
for(i = 0; i < table_size; i++) {
table[i][1] = 0; //bits
table[i][0] = -1; //codes
}
// first pass: map codes and compute auxillary table sizes
for(i = 0; i < nb_codes; i++) {
GET_DATA(n, bits, i, bits_wrap, bits_size);
GET_DATA(code, codes, i, codes_wrap, codes_size);
// we accept tables with holes
if (n <= 0)
continue;
if (!symbols)
symbol = i;
else
GET_DATA(symbol, symbols, i, symbols_wrap, symbols_size);
// if code matches the prefix, it is in the table
n -= n_prefix;
if(flags & 2)
code_prefix2= code & (n_prefix>=32 ? 0xffffffff : (1 << n_prefix)-1);
else
code_prefix2= code >> n;
if (n > 0 && code_prefix2 == code_prefix) {
if (n <= table_nb_bits) {
// no need to add another table
j = (code << (table_nb_bits - n)) & (table_size - 1);
nb = 1 << (table_nb_bits - n);
for(k = 0; k < nb; k++) {
if(flags & 2)
j = (code >> n_prefix) + (k<<n);
if (table[j][1] /*bits*/ != 0) {
error("QDM2 incorrect codes");
return -1;
}
table[j][1] = n; //bits
table[j][0] = symbol;
j++;
}
} else {
n -= table_nb_bits;
j = (code >> ((flags & 2) ? n_prefix : n)) & ((1 << table_nb_bits) - 1);
// compute table size
n1 = -table[j][1]; //bits
if (n > n1)
n1 = n;
table[j][1] = -n1; //bits
}
}
}
// second pass : fill auxillary tables recursively
for(i = 0;i < table_size; i++) {
n = table[i][1]; //bits
if (n < 0) {
n = -n;
if (n > table_nb_bits) {
n = table_nb_bits;
table[i][1] = -n; //bits
}
index = build_table(vlc, n, nb_codes,
bits, bits_wrap, bits_size,
codes, codes_wrap, codes_size,
symbols, symbols_wrap, symbols_size,
(flags & 2) ? (code_prefix | (i << n_prefix)) : ((code_prefix << table_nb_bits) | i),
n_prefix + table_nb_bits, flags);
if (index < 0)
return -1;
// note: realloc has been done, so reload tables
table = &vlc->table[table_index];
table[i][0] = index; //code
}
}
return table_index;
}
/* Build VLC decoding tables suitable for use with get_vlc().
'nb_bits' set thee decoding table size (2^nb_bits) entries. The
bigger it is, the faster is the decoding. But it should not be too
big to save memory and L1 cache. '9' is a good compromise.
'nb_codes' : number of vlcs codes
'bits' : table which gives the size (in bits) of each vlc code.
'codes' : table which gives the bit pattern of of each vlc code.
'symbols' : table which gives the values to be returned from get_vlc().
'xxx_wrap' : give the number of bytes between each entry of the
'bits' or 'codes' tables.
'xxx_size' : gives the number of bytes of each entry of the 'bits'
or 'codes' tables.
'wrap' and 'size' allows to use any memory configuration and types
(byte/word/long) to store the 'bits', 'codes', and 'symbols' tables.
'use_static' should be set to 1 for tables, which should be freed
with av_free_static(), 0 if free_vlc() will be used.
*/
void initVlcSparse(VLC *vlc, int nb_bits, int nb_codes,
const void *bits, int bits_wrap, int bits_size,
const void *codes, int codes_wrap, int codes_size,
const void *symbols, int symbols_wrap, int symbols_size) {
vlc->bits = nb_bits;
if (vlc->table_size && vlc->table_size == vlc->table_allocated) {
return;
} else if (vlc->table_size) {
error("called on a partially initialized table");
}
if (build_table(vlc, nb_bits, nb_codes,
bits, bits_wrap, bits_size,
codes, codes_wrap, codes_size,
symbols, symbols_wrap, symbols_size,
0, 0, 4 | 2) < 0) {
free(vlc->table);
return; // Error
}
if(vlc->table_size != vlc->table_allocated)
error("QDM2 needed %d had %d", vlc->table_size, vlc->table_allocated);
}
void QDM2Stream::softclipTableInit(void) {
uint16 i;
double dfl = SOFTCLIP_THRESHOLD - 32767;
float delta = 1.0 / -dfl;
for (i = 0; i < ARRAYSIZE(_softclipTable); i++)
_softclipTable[i] = SOFTCLIP_THRESHOLD - ((int)(sin((float)i * delta) * dfl) & 0x0000FFFF);
}
// random generated table
void QDM2Stream::rndTableInit(void) {
uint16 i;
uint16 j;
uint32 ldw, hdw;
// TODO: Replace Code with uint64 less version...
int64_t tmp64_1;
int64_t random_seed = 0;
float delta = 1.0 / 16384.0;
for(i = 0; i < ARRAYSIZE(_noiseTable); i++) {
random_seed = random_seed * 214013 + 2531011;
_noiseTable[i] = (delta * (float)(((int32)random_seed >> 16) & 0x00007FFF)- 1.0) * 1.3;
}
for (i = 0; i < 256; i++) {
random_seed = 81;
ldw = i;
for (j = 0; j < 5; j++) {
_randomDequantIndex[i][j] = (uint8)((ldw / random_seed) & 0xFF);
ldw = (uint32)ldw % (uint32)random_seed;
tmp64_1 = (random_seed * 0x55555556);
hdw = (uint32)(tmp64_1 >> 32);
random_seed = (int64_t)(hdw + (ldw >> 31));
}
}
for (i = 0; i < 128; i++) {
random_seed = 25;
ldw = i;
for (j = 0; j < 3; j++) {
_randomDequantType24[i][j] = (uint8)((ldw / random_seed) & 0xFF);
ldw = (uint32)ldw % (uint32)random_seed;
tmp64_1 = (random_seed * 0x66666667);
hdw = (uint32)(tmp64_1 >> 33);
random_seed = hdw + (ldw >> 31);
}
}
}
void QDM2Stream::initNoiseSamples(void) {
uint16 i;
uint32 random_seed = 0;
float delta = 1.0 / 16384.0;
for (i = 0; i < ARRAYSIZE(_noiseSamples); i++) {
random_seed = random_seed * 214013 + 2531011;
_noiseSamples[i] = (delta * (float)((random_seed >> 16) & 0x00007fff) - 1.0);
}
}
static const uint16 qdm2_vlc_offs[18] = {
0, 260, 566, 598, 894, 1166, 1230, 1294, 1678, 1950, 2214, 2278, 2310, 2570, 2834, 3124, 3448, 3838
};
void QDM2Stream::initVlc(void) {
static int16 qdm2_table[3838][2];
if (!_vlcsInitialized) {
_vlcTabLevel.table = &qdm2_table[qdm2_vlc_offs[0]];
_vlcTabLevel.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0];
_vlcTabLevel.table_size = 0;
initVlcSparse(&_vlcTabLevel, 8, 24,
vlc_tab_level_huffbits, 1, 1,
vlc_tab_level_huffcodes, 2, 2, NULL, 0, 0);
_vlcTabDiff.table = &qdm2_table[qdm2_vlc_offs[1]];
_vlcTabDiff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1];
_vlcTabDiff.table_size = 0;
initVlcSparse(&_vlcTabDiff, 8, 37,
vlc_tab_diff_huffbits, 1, 1,
vlc_tab_diff_huffcodes, 2, 2, NULL, 0, 0);
_vlcTabRun.table = &qdm2_table[qdm2_vlc_offs[2]];
_vlcTabRun.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2];
_vlcTabRun.table_size = 0;
initVlcSparse(&_vlcTabRun, 5, 6,
vlc_tab_run_huffbits, 1, 1,
vlc_tab_run_huffcodes, 1, 1, NULL, 0, 0);
_fftLevelExpAltVlc.table = &qdm2_table[qdm2_vlc_offs[3]];
_fftLevelExpAltVlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3];
_fftLevelExpAltVlc.table_size = 0;
initVlcSparse(&_fftLevelExpAltVlc, 8, 28,
fft_level_exp_alt_huffbits, 1, 1,
fft_level_exp_alt_huffcodes, 2, 2, NULL, 0, 0);
_fftLevelExpVlc.table = &qdm2_table[qdm2_vlc_offs[4]];
_fftLevelExpVlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4];
_fftLevelExpVlc.table_size = 0;
initVlcSparse(&_fftLevelExpVlc, 8, 20,
fft_level_exp_huffbits, 1, 1,
fft_level_exp_huffcodes, 2, 2, NULL, 0, 0);
_fftStereoExpVlc.table = &qdm2_table[qdm2_vlc_offs[5]];
_fftStereoExpVlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5];
_fftStereoExpVlc.table_size = 0;
initVlcSparse(&_fftStereoExpVlc, 6, 7,
fft_stereo_exp_huffbits, 1, 1,
fft_stereo_exp_huffcodes, 1, 1, NULL, 0, 0);
_fftStereoPhaseVlc.table = &qdm2_table[qdm2_vlc_offs[6]];
_fftStereoPhaseVlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6];
_fftStereoPhaseVlc.table_size = 0;
initVlcSparse(&_fftStereoPhaseVlc, 6, 9,
fft_stereo_phase_huffbits, 1, 1,
fft_stereo_phase_huffcodes, 1, 1, NULL, 0, 0);
_vlcTabToneLevelIdxHi1.table = &qdm2_table[qdm2_vlc_offs[7]];
_vlcTabToneLevelIdxHi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7];
_vlcTabToneLevelIdxHi1.table_size = 0;
initVlcSparse(&_vlcTabToneLevelIdxHi1, 8, 20,
vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, NULL, 0, 0);
_vlcTabToneLevelIdxMid.table = &qdm2_table[qdm2_vlc_offs[8]];
_vlcTabToneLevelIdxMid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8];
_vlcTabToneLevelIdxMid.table_size = 0;
initVlcSparse(&_vlcTabToneLevelIdxMid, 8, 24,
vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, NULL, 0, 0);
_vlcTabToneLevelIdxHi2.table = &qdm2_table[qdm2_vlc_offs[9]];
_vlcTabToneLevelIdxHi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9];
_vlcTabToneLevelIdxHi2.table_size = 0;
initVlcSparse(&_vlcTabToneLevelIdxHi2, 8, 24,
vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, NULL, 0, 0);
_vlcTabType30.table = &qdm2_table[qdm2_vlc_offs[10]];
_vlcTabType30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10];
_vlcTabType30.table_size = 0;
initVlcSparse(&_vlcTabType30, 6, 9,
vlc_tab_type30_huffbits, 1, 1,
vlc_tab_type30_huffcodes, 1, 1, NULL, 0, 0);
_vlcTabType34.table = &qdm2_table[qdm2_vlc_offs[11]];
_vlcTabType34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11];
_vlcTabType34.table_size = 0;
initVlcSparse(&_vlcTabType34, 5, 10,
vlc_tab_type34_huffbits, 1, 1,
vlc_tab_type34_huffcodes, 1, 1, NULL, 0, 0);
_vlcTabFftToneOffset[0].table = &qdm2_table[qdm2_vlc_offs[12]];
_vlcTabFftToneOffset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12];
_vlcTabFftToneOffset[0].table_size = 0;
initVlcSparse(&_vlcTabFftToneOffset[0], 8, 23,
vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, NULL, 0, 0);
_vlcTabFftToneOffset[1].table = &qdm2_table[qdm2_vlc_offs[13]];
_vlcTabFftToneOffset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13];
_vlcTabFftToneOffset[1].table_size = 0;
initVlcSparse(&_vlcTabFftToneOffset[1], 8, 28,
vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, NULL, 0, 0);
_vlcTabFftToneOffset[2].table = &qdm2_table[qdm2_vlc_offs[14]];
_vlcTabFftToneOffset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14];
_vlcTabFftToneOffset[2].table_size = 0;
initVlcSparse(&_vlcTabFftToneOffset[2], 8, 32,
vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, NULL, 0, 0);
_vlcTabFftToneOffset[3].table = &qdm2_table[qdm2_vlc_offs[15]];
_vlcTabFftToneOffset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15];
_vlcTabFftToneOffset[3].table_size = 0;
initVlcSparse(&_vlcTabFftToneOffset[3], 8, 35,
vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, NULL, 0, 0);
_vlcTabFftToneOffset[4].table = &qdm2_table[qdm2_vlc_offs[16]];
_vlcTabFftToneOffset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16];
_vlcTabFftToneOffset[4].table_size = 0;
initVlcSparse(&_vlcTabFftToneOffset[4], 8, 38,
vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, NULL, 0, 0);
_vlcsInitialized = true;
}
}
QDM2Stream::QDM2Stream(Common::SeekableReadStream *extraData, DisposeAfterUse::Flag disposeExtraData) {
uint32 tmp;
int tmp_val;
int i;
debug(1, "QDM2Stream::QDM2Stream() Call");
_compressedData = NULL;
_subPacket = 0;
_superBlockStart = 0;
memset(_quantizedCoeffs, 0, sizeof(_quantizedCoeffs));
memset(_fftLevelExp, 0, sizeof(_fftLevelExp));
_noiseIdx = 0;
memset(_fftCoefsMinIndex, 0, sizeof(_fftCoefsMinIndex));
memset(_fftCoefsMaxIndex, 0, sizeof(_fftCoefsMaxIndex));
_fftToneStart = 0;
_fftToneEnd = 0;
for(i = 0; i < ARRAYSIZE(_subPacketListA); i++) {
_subPacketListA[i].packet = NULL;
_subPacketListA[i].next = NULL;
}
_subPacketsB = 0;
for(i = 0; i < ARRAYSIZE(_subPacketListB); i++) {
_subPacketListB[i].packet = NULL;
_subPacketListB[i].next = NULL;
}
for(i = 0; i < ARRAYSIZE(_subPacketListC); i++) {
_subPacketListC[i].packet = NULL;
_subPacketListC[i].next = NULL;
}
for(i = 0; i < ARRAYSIZE(_subPacketListD); i++) {
_subPacketListD[i].packet = NULL;
_subPacketListD[i].next = NULL;
}
memset(_synthBuf, 0, sizeof(_synthBuf));
memset(_synthBufOffset, 0, sizeof(_synthBufOffset));
memset(_sbSamples, 0, sizeof(_sbSamples));
memset(_outputBuffer, 0, sizeof(_outputBuffer));
_vlcsInitialized = false;
_superblocktype_2_3 = 0;
_hasErrors = false;
// The QDM2 "extra data" is really just an amalgam of three QuickTime
// atoms needed to correctly set up the decoder.
// Rewind extraData stream from any previous calls
extraData->seek(0, SEEK_SET);
// First, the frma atom
uint32 frmaSize = extraData->readUint32BE();
if (frmaSize != 12)
error("Invalid QDM2 frma atom");
if (extraData->readUint32BE() != MKTAG('f', 'r', 'm', 'a'))
error("Failed to find frma atom for QDM2");
uint32 version = extraData->readUint32BE();
if (version == MKTAG('Q', 'D', 'M', 'C'))
error("Unhandled QDMC sound");
else if (version != MKTAG('Q', 'D', 'M', '2'))
error("Failed to find QDM2 tag in frma atom");
// Second, the QDCA atom
uint32 qdcaSize = extraData->readUint32BE();
if (qdcaSize > (uint32)(extraData->size() - extraData->pos()))
error("Invalid QDM2 QDCA atom");
if (extraData->readUint32BE() != MKTAG('Q', 'D', 'C', 'A'))
error("Failed to find QDCA atom for QDM2");
extraData->readUint32BE(); // unknown
_channels = extraData->readUint32BE();
_sampleRate = extraData->readUint32BE();
_bitRate = extraData->readUint32BE();
_blockSize = extraData->readUint32BE();
_frameSize = extraData->readUint32BE();
_packetSize = extraData->readUint32BE();
// Third, we don't care about the QDCP atom
_fftOrder = Common::intLog2(_frameSize) + 1;
_fftFrameSize = 2 * _frameSize; // complex has two floats
// something like max decodable tones
_groupOrder = Common::intLog2(_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);
_rdft = new Common::RDFT(_fftOrder, Common::RDFT::IDFT_C2R);
initVlc();
ff_mpa_synth_init(ff_mpa_synth_window);
softclipTableInit();
rndTableInit();
initNoiseSamples();
_compressedData = new uint8[_packetSize];
if (disposeExtraData == DisposeAfterUse::YES)
delete extraData;
}
QDM2Stream::~QDM2Stream() {
delete _rdft;
delete[] _compressedData;
}
static int qdm2_get_vlc(Common::BitStream *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 = gb->getBits(gb->getBits(3) + 1);
// stage-3, optional
if (flag) {
int tmp = vlc_stage3_values[value];
if ((value & ~3) > 0)
tmp += gb->getBits(value >> 2);
value = tmp;
}
return value;
}
static int qdm2_get_se_vlc(VLC *vlc, Common::BitStream *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);
}
/**
* 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(Common::BitStream *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 = ((length - gb->pos()) >= 1) ? gb->getBit() : 0;
if (joined_stereo) {
if ((length - gb->pos()) >= 16)
for (j = 0; j < 16; j++)
sign_bits[j] = gb->getBit();
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 = ((length - gb->pos()) >= 1) ? gb->getBit() : 0;
type34_predictor = 0.0;
type34_first = 1;
for (j = 0; j < 128; ) {
switch (_codingMethod[ch][sb][j / 2]) {
case 8:
if ((length - gb->pos()) >= 10) {
if (zero_encoding) {
for (k = 0; k < 5; k++) {
if ((j + 2 * k) >= 128)
break;
samples[2 * k] = gb->getBit() ? dequant_1bit[joined_stereo][2 * gb->getBit()] : 0;
}
} else {
n = gb->getBits(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 ((length - gb->pos()) >= 1) {
double f = 0.81;
if (gb->getBit())
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 ((length - gb->pos()) >= 10) {
if (zero_encoding) {
for (k = 0; k < 5; k++) {
if ((j + k) >= 128)
break;
samples[k] = (gb->getBit() == 0) ? 0 : dequant_1bit[joined_stereo][2 * gb->getBit()];
}
} else {
n = gb->getBits(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 ((length - gb->pos()) >= 7) {
n = gb->getBits(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 ((length - gb->pos()) >= 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 ((length - gb->pos()) >= 7) {
if (type34_first) {
type34_div = (float)(1 << gb->getBits(2));
samples[0] = ((float)gb->getBits(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, Common::BitStream *gb, int length) {
int i, k, run, level, diff;
if ((length - gb->pos()) < 16)
return;
level = qdm2_get_vlc(gb, &_vlcTabLevel, 0, 2);
quantized_coeffs[0] = level;
for (i = 0; i < 7; ) {
if ((length - gb->pos()) < 16)
break;
run = qdm2_get_vlc(gb, &_vlcTabRun, 0, 1) + 1;
if ((length - gb->pos()) < 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(Common::BitStream *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 ((length - gb->pos()) < 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 ((length - gb->pos()) < 1)
break;
if (gb->getBit()) {
for (k=0; k < 8; k++) {
if ((length - gb->pos()) < 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 ((length - gb->pos()) < 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 ((length - gb->pos()) < 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) {
int i, j, k, n, ch, run, level, diff;
Common::MemoryReadStream d(node->packet->data, node->packet->size*8);
Common::BitStream32LELSB gb(&d);
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) {
Common::MemoryReadStream d(((node == NULL) ? _emptyBuffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
Common::BitStream32LELSB gb(&d);
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) {
Common::MemoryReadStream d(((node == NULL) ? _emptyBuffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
Common::BitStream32LELSB gb(&d);
if (length >= 32) {
int c = gb.getBits(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) {
Common::MemoryReadStream d(((node == NULL) ? _emptyBuffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
Common::BitStream32LELSB gb(&d);
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) {
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]
Common::MemoryReadStream *d = new Common::MemoryReadStream(_compressedData, _packetSize*8);
Common::BitStream *gb = new Common::BitStream32LELSB(d);
//qdm2_decode_sub_packet_header
header.type = gb->getBits(8);
if (header.type == 0) {
header.size = 0;
header.data = NULL;
} else {
header.size = gb->getBits(8);
if (header.type & 0x80) {
header.size <<= 8;
header.size |= gb->getBits(8);
header.type &= 0x7f;
}
if (header.type == 0x7f)
header.type |= (gb->getBits(8) << 8);
header.data = &_compressedData[gb->pos() / 8];
}
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 - gb->pos() / 8);
delete gb;
delete d;
d = new Common::MemoryReadStream(header.data, header.size*8);
gb = new Common::BitStream32LELSB(d);
if (header.type == 2 || header.type == 4 || header.type == 5) {
int csum = 257 * gb->getBits(8) + 2 * gb->getBits(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
delete gb;
delete d;
d = new Common::MemoryReadStream(header.data, header.size*8);
gb = new Common::BitStream32LELSB(d);
gb->skip(next_index*8);
if (next_index >= header.size)
break;
}
// decode subpacket
packet = &_subPackets[i];
//qdm2_decode_sub_packet_header
packet->type = gb->getBits(8);
if (packet->type == 0) {
packet->size = 0;
packet->data = NULL;
} else {
packet->size = gb->getBits(8);
if (packet->type & 0x80) {
packet->size <<= 8;
packet->size |= gb->getBits(8);
packet->type &= 0x7f;
}
if (packet->type == 0x7f)
packet->type |= (gb->getBits(8) << 8);
packet->data = &header.data[gb->pos() / 8];
}
next_index = packet->size + gb->pos() / 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");
delete gb;
delete d;
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] = gb->getBits(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");
delete gb;
delete d;
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);
}
// ****************************************************************
delete gb;
delete d;
}
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, Common::BitStream *gb, int b) {
int channel, stereo, phase, exp;
int local_int_4, local_int_8, stereo_phase, local_int_10;
int local_int_14, stereo_exp, local_int_20, local_int_28;
int n, offset;
local_int_4 = 0;
local_int_28 = 0;
local_int_20 = 2;
local_int_8 = (4 - duration);
local_int_10 = 1 << (_groupOrder - duration - 1);
offset = 1;
while (1) {
if (_superblocktype_2_3) {
while ((n = qdm2_get_vlc(gb, &_vlcTabFftToneOffset[local_int_8], 1, 2)) < 2) {
offset = 1;
if (n == 0) {
local_int_4 += local_int_10;
local_int_28 += (1 << local_int_8);
} else {
local_int_4 += 8*local_int_10;
local_int_28 += (8 << local_int_8);
}
}
offset += (n - 2);
} else {
offset += qdm2_get_vlc(gb, &_vlcTabFftToneOffset[local_int_8], 1, 2);
while (offset >= (local_int_10 - 1)) {
offset += (1 - (local_int_10 - 1));
local_int_4 += local_int_10;
local_int_28 += (1 << local_int_8);
}
}
if (local_int_4 >= _blockSize)
return;
local_int_14 = (offset >> local_int_8);
if (_channels > 1) {
channel = gb->getBit();
stereo = gb->getBit();
} 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 = gb->getBits(3);
stereo_exp = 0;
stereo_phase = 0;
if (stereo) {
stereo_exp = (exp - qdm2_get_vlc(gb, &_fftStereoExpVlc, 0, 1));
stereo_phase = (phase - qdm2_get_vlc(gb, &_fftStereoPhaseVlc, 0, 1));
if (stereo_phase < 0)
stereo_phase += 8;
}
if (_frequencyRange > (local_int_14 + 1)) {
int sub_packet = (local_int_20 + local_int_28);
qdm2_fft_init_coefficient(sub_packet, offset, duration, channel, exp, phase);
if (stereo)
qdm2_fft_init_coefficient(sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
}
offset++;
}
}
void QDM2Stream::qdm2_decode_fft_packets(void) {
int i, j, min, max, value, type, unknown_flag;
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
Common::MemoryReadStream d(packet->data, packet->size*8);
Common::BitStream32LELSB gb(&d);
if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
unknown_flag = 1;
else
unknown_flag = 0;
type = packet->type;
if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
int duration = _subSampling + 5 - (type & 15);
if (duration >= 0 && duration < 4) { // TODO: Should be <= 4?
qdm2_fft_decode_tones(duration, &gb, unknown_flag);
}
} else if (type == 31) {
for (j=0; j < 4; j++) {
qdm2_fft_decode_tones(j, &gb, unknown_flag);
}
} else if (type == 46) {
for (j=0; j < 6; j++)
_fftLevelExp[j] = gb.getBits(6);
for (j=0; j < 4; j++) {
qdm2_fft_decode_tones(j, &gb, unknown_flag);
}
}
} // Loop on B packets
// calculate maximum indexes for FFT coefficients
for (i = 0, j = -1; i < 5; i++)
if (_fftCoefsMinIndex[i] >= 0) {
if (j >= 0)
_fftCoefsMaxIndex[j] = _fftCoefsMinIndex[i];
j = i;
}
if (j >= 0)
_fftCoefsMaxIndex[j] = _fftCoefsIndex;
}
void QDM2Stream::qdm2_fft_generate_tone(FFTTone *tone)
{
float level, f[6];
int i;
QDM2Complex c;
const double iscale = 2.0 * M_PI / 512.0;
tone->phase += tone->phase_shift;
// calculate current level (maximum amplitude) of tone
level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
c.im = level * sin(tone->phase*iscale);
c.re = level * cos(tone->phase*iscale);
// generate FFT coefficients for tone
if (tone->duration >= 3 || tone->cutoff >= 3) {
tone->complex[0].im += c.im;
tone->complex[0].re += c.re;
tone->complex[1].im -= c.im;
tone->complex[1].re -= c.re;
} else {
f[1] = -tone->table[4];
f[0] = tone->table[3] - tone->table[0];
f[2] = 1.0 - tone->table[2] - tone->table[3];
f[3] = tone->table[1] + tone->table[4] - 1.0;
f[4] = tone->table[0] - tone->table[1];
f[5] = tone->table[2];
for (i = 0; i < 2; i++) {
tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i];
tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
}
for (i = 0; i < 4; i++) {
tone->complex[i].re += c.re * f[i+2];
tone->complex[i].im += c.im * f[i+2];
}
}
// copy the tone if it has not yet died out
if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
memcpy(&_fftTones[_fftToneEnd], tone, sizeof(FFTTone));
_fftToneEnd = (_fftToneEnd + 1) % 1000;
}
}
void QDM2Stream::qdm2_fft_tone_synthesizer(uint8 sub_packet) {
int i, j, ch;
const double iscale = 0.25 * M_PI;
for (ch = 0; ch < _channels; ch++) {
memset(_fft.complex[ch], 0, _frameSize * sizeof(QDM2Complex));
}
// apply FFT tones with duration 4 (1 FFT period)
if (_fftCoefsMinIndex[4] >= 0)
for (i = _fftCoefsMinIndex[4]; i < _fftCoefsMaxIndex[4]; i++) {
float level;
QDM2Complex c;
if (_fftCoefs[i].sub_packet != sub_packet)
break;
ch = (_channels == 1) ? 0 : _fftCoefs[i].channel;
level = (_fftCoefs[i].exp < 0) ? 0.0 : fft_tone_level_table[_superblocktype_2_3 ? 0 : 1][_fftCoefs[i].exp & 63];
c.re = level * cos(_fftCoefs[i].phase * iscale);
c.im = level * sin(_fftCoefs[i].phase * iscale);
_fft.complex[ch][_fftCoefs[i].offset + 0].re += c.re;
_fft.complex[ch][_fftCoefs[i].offset + 0].im += c.im;
_fft.complex[ch][_fftCoefs[i].offset + 1].re -= c.re;
_fft.complex[ch][_fftCoefs[i].offset + 1].im -= c.im;
}
// generate existing FFT tones
for (i = _fftToneEnd; i != _fftToneStart; ) {
qdm2_fft_generate_tone(&_fftTones[_fftToneStart]);
_fftToneStart = (_fftToneStart + 1) % 1000;
}
// create and generate new FFT tones with duration 0 (long) to 3 (short)
for (i = 0; i < 4; i++)
if (_fftCoefsMinIndex[i] >= 0) {
for (j = _fftCoefsMinIndex[i]; j < _fftCoefsMaxIndex[i]; j++) {
int offset, four_i;
FFTTone tone;
if (_fftCoefs[j].sub_packet != sub_packet)
break;
four_i = (4 - i);
offset = _fftCoefs[j].offset >> four_i;
ch = (_channels == 1) ? 0 : _fftCoefs[j].channel;
if (offset < _frequencyRange) {
if (offset < 2)
tone.cutoff = offset;
else
tone.cutoff = (offset >= 60) ? 3 : 2;
tone.level = (_fftCoefs[j].exp < 0) ? 0.0 : fft_tone_level_table[_superblocktype_2_3 ? 0 : 1][_fftCoefs[j].exp & 63];
tone.complex = &_fft.complex[ch][offset];
tone.table = fft_tone_sample_table[i][_fftCoefs[j].offset - (offset << four_i)];
tone.phase = 64 * _fftCoefs[j].phase - (offset << 8) - 128;
tone.phase_shift = (2 * _fftCoefs[j].offset + 1) << (7 - four_i);
tone.duration = i;
tone.time_index = 0;
qdm2_fft_generate_tone(&tone);
}
}
_fftCoefsMinIndex[i] = j;
}
}
void QDM2Stream::qdm2_calculate_fft(int channel) {
_fft.complex[channel][0].re *= 2.0f;
_fft.complex[channel][0].im = 0.0f;
_rdft->calc((float *)_fft.complex[channel]);
// add samples to output buffer
for (int i = 0; i < ((_fftFrameSize + 15) & ~15); i++)
_outputBuffer[_channels * i + channel] += ((float *) _fft.complex[channel])[i];
}
/**
* @param index subpacket number
*/
void QDM2Stream::qdm2_synthesis_filter(uint8 index)
{
int16 samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
int i, k, ch, sb_used, sub_sampling, dither_state = 0;
// copy sb_samples
sb_used = QDM2_SB_USED(_subSampling);
for (ch = 0; ch < _channels; ch++)
for (i = 0; i < 8; i++)
for (k = sb_used; k < 32; k++)
_sbSamples[ch][(8 * index) + i][k] = 0;
for (ch = 0; ch < _channels; ch++) {
int16 *samples_ptr = samples + ch;
for (i = 0; i < 8; i++) {
ff_mpa_synth_filter(_synthBuf[ch], &(_synthBufOffset[ch]),
ff_mpa_synth_window, &dither_state,
samples_ptr, _channels,
_sbSamples[ch][(8 * index) + i]);
samples_ptr += 32 * _channels;
}
}
// add samples to output buffer
sub_sampling = (4 >> _subSampling);
for (ch = 0; ch < _channels; ch++)
for (i = 0; i < _sFrameSize; i++)
_outputBuffer[_channels * i + ch] += (float)(samples[_channels * sub_sampling * i + ch] >> (sizeof(int16)*8-16));
}
bool QDM2Stream::qdm2_decodeFrame(Common::SeekableReadStream &in, QueuingAudioStream *audioStream) {
debug(1, "QDM2Stream::qdm2_decodeFrame in.pos(): %d in.size(): %d", in.pos(), in.size());
int ch, i;
const int frame_size = (_sFrameSize * _channels);
// If we're in any packet but the first, seek back to the first
if (_subPacket == 0)
_superBlockStart = in.pos();
else
in.seek(_superBlockStart);
// select input buffer
if (in.eos() || in.pos() >= in.size()) {
debug(1, "QDM2Stream::qdm2_decodeFrame End of Input Stream");
return false;
}
if ((in.size() - in.pos()) < _packetSize) {
debug(1, "QDM2Stream::qdm2_decodeFrame Insufficient Packet Data in Input Stream Found: %d Need: %d", in.size() - in.pos(), _packetSize);
return false;
}
if (!in.eos()) {
in.read(_compressedData, _packetSize);
debug(1, "QDM2Stream::qdm2_decodeFrame constructed input data");
}
// copy old block, clear new block of output samples
memmove(_outputBuffer, &_outputBuffer[frame_size], frame_size * sizeof(float));
memset(&_outputBuffer[frame_size], 0, frame_size * sizeof(float));
debug(1, "QDM2Stream::qdm2_decodeFrame cleared outputBuffer");
if (!in.eos()) {
// decode block of QDM2 compressed data
debug(1, "QDM2Stream::qdm2_decodeFrame decode block of QDM2 compressed data");
if (_subPacket == 0) {
_hasErrors = false; // reset it for a new super block
debug(1, "QDM2 : Superblock follows");
qdm2_decode_super_block();
}
// parse subpackets
debug(1, "QDM2Stream::qdm2_decodeFrame parse subpackets");
if (!_hasErrors) {
if (_subPacket == 2) {
debug(1, "QDM2Stream::qdm2_decodeFrame qdm2_decode_fft_packets()");
qdm2_decode_fft_packets();
}
debug(1, "QDM2Stream::qdm2_decodeFrame qdm2_fft_tone_synthesizer(%d)", _subPacket);
qdm2_fft_tone_synthesizer(_subPacket);
}
// sound synthesis stage 1 (FFT)
debug(1, "QDM2Stream::qdm2_decodeFrame sound synthesis stage 1 (FFT)");
for (ch = 0; ch < _channels; ch++) {
qdm2_calculate_fft(ch);
if (!_hasErrors && _subPacketListC[0].packet != NULL) {
error("QDM2 : has errors, and C list is not empty");
return false;
}
}
// 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");
}
if (frame_size == 0)
return false;
// Prepare a buffer for queuing
uint16 *outputBuffer = (uint16 *)malloc(frame_size * 2);
for (i = 0; i < frame_size; i++) {
int value = (int)_outputBuffer[i];
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];
outputBuffer[i] = value;
}
// Queue the translated buffer to our stream
byte flags = FLAG_16BITS;
if (_channels == 2)
flags |= FLAG_STEREO;
#ifdef SCUMM_LITTLE_ENDIAN
flags |= FLAG_LITTLE_ENDIAN;
#endif
audioStream->queueBuffer((byte *)outputBuffer, frame_size * 2, DisposeAfterUse::YES, flags);
return true;
}
AudioStream *QDM2Stream::decodeFrame(Common::SeekableReadStream &stream) {
QueuingAudioStream *audioStream = makeQueuingAudioStream(_sampleRate, _channels == 2);
while (qdm2_decodeFrame(stream, audioStream))
;
audioStream->finish();
return audioStream;
}
Codec *makeQDM2Decoder(Common::SeekableReadStream *extraData, DisposeAfterUse::Flag disposeExtraData) {
return new QDM2Stream(extraData, disposeExtraData);
}
} // End of namespace Audio
#endif
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