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Diffstat (limited to 'sound/decoders/qdm2.h')
| -rw-r--r-- | sound/decoders/qdm2.h | 294 |
1 files changed, 294 insertions, 0 deletions
diff --git a/sound/decoders/qdm2.h b/sound/decoders/qdm2.h new file mode 100644 index 0000000000..77208aec52 --- /dev/null +++ b/sound/decoders/qdm2.h @@ -0,0 +1,294 @@ +/* ScummVM - Graphic Adventure Engine + * + * ScummVM is the legal property of its developers, whose names + * are too numerous to list here. Please refer to the COPYRIGHT + * file distributed with this source distribution. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA. + * + * $URL$ + * $Id$ + * + */ + +// Only compile if Mohawk is enabled or if we're building dynamic modules +#if defined(ENABLE_MOHAWK) || defined(DYNAMIC_MODULES) + +#ifndef SOUND_QDM2_H +#define SOUND_QDM2_H + +#include "sound/audiostream.h" +#include "common/array.h" +#include "common/stream.h" + +namespace Audio { + +enum { + SOFTCLIP_THRESHOLD = 27600, + HARDCLIP_THRESHOLD = 35716, + MPA_MAX_CHANNELS = 2, + MPA_FRAME_SIZE = 1152, + FF_INPUT_BUFFER_PADDING_SIZE = 8 +}; + +typedef int8 sb_int8_array[2][30][64]; + +/* bit input */ +/* buffer, buffer_end and size_in_bits must be present and used by every reader */ +struct GetBitContext { + const uint8 *buffer, *bufferEnd; + int index; + int sizeInBits; +}; + +struct QDM2SubPacket { + int type; + unsigned int size; + const uint8 *data; // pointer to subpacket data (points to input data buffer, it's not a private copy) +}; + +struct QDM2SubPNode { + QDM2SubPacket *packet; + struct QDM2SubPNode *next; // pointer to next packet in the list, NULL if leaf node +}; + +struct QDM2Complex { + float re; + float im; +}; + +struct FFTTone { + float level; + QDM2Complex *complex; + const float *table; + int phase; + int phase_shift; + int duration; + short time_index; + short cutoff; +}; + +struct FFTCoefficient { + int16 sub_packet; + uint8 channel; + int16 offset; + int16 exp; + uint8 phase; +}; + +struct VLC { + int32 bits; + int16 (*table)[2]; // code, bits + int32 table_size; + int32 table_allocated; +}; + +#include "common/pack-start.h" +struct QDM2FFT { + QDM2Complex complex[MPA_MAX_CHANNELS][256]; +} PACKED_STRUCT; +#include "common/pack-end.h" + +enum RDFTransformType { + RDFT, + IRDFT, + RIDFT, + IRIDFT +}; + +struct FFTComplex { + float re, im; +}; + +struct FFTContext { + int nbits; + int inverse; + uint16 *revtab; + FFTComplex *exptab; + FFTComplex *tmpBuf; + int mdctSize; // size of MDCT (i.e. number of input data * 2) + int mdctBits; // n = 2^nbits + // pre/post rotation tables + float *tcos; + float *tsin; + void (*fftPermute)(struct FFTContext *s, FFTComplex *z); + void (*fftCalc)(struct FFTContext *s, FFTComplex *z); + void (*imdctCalc)(struct FFTContext *s, float *output, const float *input); + void (*imdctHalf)(struct FFTContext *s, float *output, const float *input); + void (*mdctCalc)(struct FFTContext *s, float *output, const float *input); + int splitRadix; + int permutation; +}; + +enum { + FF_MDCT_PERM_NONE = 0, + FF_MDCT_PERM_INTERLEAVE = 1 +}; + +struct RDFTContext { + int nbits; + int inverse; + int signConvention; + + // pre/post rotation tables + float *tcos; + float *tsin; + FFTContext fft; +}; + +class QDM2Stream : public Audio::AudioStream { +public: + QDM2Stream(Common::SeekableReadStream *stream, Common::SeekableReadStream *extraData); + ~QDM2Stream(); + + bool isStereo() const { return _channels == 2; } + bool endOfData() const { return ((_stream->pos() == _stream->size()) && (_outputSamples.size() == 0)); } + int getRate() const { return _sampleRate; } + int readBuffer(int16 *buffer, const int numSamples); + +private: + Common::SeekableReadStream *_stream; + + // Parameters from codec header, do not change during playback + uint8 _channels; + uint16 _sampleRate; + uint16 _bitRate; + uint16 _blockSize; // Group + uint16 _frameSize; // FFT + uint16 _packetSize; // Checksum + + // Parameters built from header parameters, do not change during playback + int _groupOrder; // order of frame group + int _fftOrder; // order of FFT (actually fft order+1) + int _fftFrameSize; // size of fft frame, in components (1 comples = re + im) + int _sFrameSize; // size of data frame + int _frequencyRange; + int _subSampling; // subsampling: 0=25%, 1=50%, 2=100% */ + int _coeffPerSbSelect; // selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2 + int _cmTableSelect; // selector for "coding method" tables. Can be 0, 1 (from init: 0-4) + + // Packets and packet lists + QDM2SubPacket _subPackets[16]; // the packets themselves + QDM2SubPNode _subPacketListA[16]; // list of all packets + QDM2SubPNode _subPacketListB[16]; // FFT packets B are on list + int _subPacketsB; // number of packets on 'B' list + QDM2SubPNode _subPacketListC[16]; // packets with errors? + QDM2SubPNode _subPacketListD[16]; // DCT packets + + // FFT and tones + FFTTone _fftTones[1000]; + int _fftToneStart; + int _fftToneEnd; + FFTCoefficient _fftCoefs[1000]; + int _fftCoefsIndex; + int _fftCoefsMinIndex[5]; + int _fftCoefsMaxIndex[5]; + int _fftLevelExp[6]; + //RDFTContext _rdftCtx; + QDM2FFT _fft; + + // I/O data + uint8 *_compressedData; + float _outputBuffer[1024]; + Common::Array<int16> _outputSamples; + + // Synthesis filter + int16 ff_mpa_synth_window[512]; + int16 _synthBuf[MPA_MAX_CHANNELS][512*2]; + int _synthBufOffset[MPA_MAX_CHANNELS]; + int32 _sbSamples[MPA_MAX_CHANNELS][128][32]; + + // Mixed temporary data used in decoding + float _toneLevel[MPA_MAX_CHANNELS][30][64]; + int8 _codingMethod[MPA_MAX_CHANNELS][30][64]; + int8 _quantizedCoeffs[MPA_MAX_CHANNELS][10][8]; + int8 _toneLevelIdxBase[MPA_MAX_CHANNELS][30][8]; + int8 _toneLevelIdxHi1[MPA_MAX_CHANNELS][3][8][8]; + int8 _toneLevelIdxMid[MPA_MAX_CHANNELS][26][8]; + int8 _toneLevelIdxHi2[MPA_MAX_CHANNELS][26]; + int8 _toneLevelIdx[MPA_MAX_CHANNELS][30][64]; + int8 _toneLevelIdxTemp[MPA_MAX_CHANNELS][30][64]; + + // Flags + bool _hasErrors; // packet has errors + int _superblocktype_2_3; // select fft tables and some algorithm based on superblock type + int _doSynthFilter; // used to perform or skip synthesis filter + + uint8 _subPacket; // 0 to 15 + int _noiseIdx; // index for dithering noise table + + byte _emptyBuffer[FF_INPUT_BUFFER_PADDING_SIZE]; + + VLC _vlcTabLevel; + VLC _vlcTabDiff; + VLC _vlcTabRun; + VLC _fftLevelExpAltVlc; + VLC _fftLevelExpVlc; + VLC _fftStereoExpVlc; + VLC _fftStereoPhaseVlc; + VLC _vlcTabToneLevelIdxHi1; + VLC _vlcTabToneLevelIdxMid; + VLC _vlcTabToneLevelIdxHi2; + VLC _vlcTabType30; + VLC _vlcTabType34; + VLC _vlcTabFftToneOffset[5]; + bool _vlcsInitialized; + void initVlc(void); + + uint16 _softclipTable[HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1]; + void softclipTableInit(void); + + float _noiseTable[4096]; + byte _randomDequantIndex[256][5]; + byte _randomDequantType24[128][3]; + void rndTableInit(void); + + float _noiseSamples[128]; + void initNoiseSamples(void); + + RDFTContext _rdftCtx; + + void average_quantized_coeffs(void); + void build_sb_samples_from_noise(int sb); + void fix_coding_method_array(int sb, int channels, sb_int8_array coding_method); + void fill_tone_level_array(int flag); + void fill_coding_method_array(sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp, + sb_int8_array coding_method, int nb_channels, + int c, int superblocktype_2_3, int cm_table_select); + void synthfilt_build_sb_samples(GetBitContext *gb, int length, int sb_min, int sb_max); + void init_quantized_coeffs_elem0(int8 *quantized_coeffs, GetBitContext *gb, int length); + void init_tone_level_dequantization(GetBitContext *gb, int length); + void process_subpacket_9(QDM2SubPNode *node); + void process_subpacket_10(QDM2SubPNode *node, int length); + void process_subpacket_11(QDM2SubPNode *node, int length); + void process_subpacket_12(QDM2SubPNode *node, int length); + void process_synthesis_subpackets(QDM2SubPNode *list); + void qdm2_decode_super_block(void); + void qdm2_fft_init_coefficient(int sub_packet, int offset, int duration, + int channel, int exp, int phase); + void qdm2_fft_decode_tones(int duration, GetBitContext *gb, int b); + void qdm2_decode_fft_packets(void); + void qdm2_fft_generate_tone(FFTTone *tone); + void qdm2_fft_tone_synthesizer(uint8 sub_packet); + void qdm2_calculate_fft(int channel); + void qdm2_synthesis_filter(uint8 index); + int qdm2_decodeFrame(Common::SeekableReadStream *in); +}; + +} // End of namespace Audio + +#endif // SOUND_QDM2_H + +#endif // Mohawk/Plugins guard |