/* 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)) ; return audioStream; } Codec *makeQDM2Decoder(Common::SeekableReadStream *extraData, DisposeAfterUse::Flag disposeExtraData) { return new QDM2Stream(extraData, disposeExtraData); } } // End of namespace Audio #endif