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| author | Matthew Hoops | 2010-05-23 21:41:13 +0000 |
|---|---|---|
| committer | Matthew Hoops | 2010-05-23 21:41:13 +0000 |
| commit | b3bd797e019c20de1d4bfdac131e6e3e2c45860a (patch) | |
| tree | 0d0b6fbb94dd2bdf8bb2fbf7f8eae9d92183b4d7 /sound | |
| parent | 5e90f66edc816931996a72b8fca8b01a96aff31a (diff) | |
| download | scummvm-rg350-b3bd797e019c20de1d4bfdac131e6e3e2c45860a.tar.gz scummvm-rg350-b3bd797e019c20de1d4bfdac131e6e3e2c45860a.tar.bz2 scummvm-rg350-b3bd797e019c20de1d4bfdac131e6e3e2c45860a.zip | |
Move the QDM2 code to the graphics module, removing the cyclic dependency.
svn-id: r49171
Diffstat (limited to 'sound')
| -rw-r--r-- | sound/decoders/qdm2.cpp | 3327 | ||||
| -rw-r--r-- | sound/decoders/qdm2.h | 51 | ||||
| -rw-r--r-- | sound/decoders/qdm2data.h | 531 | ||||
| -rw-r--r-- | sound/module.mk | 1 |
4 files changed, 0 insertions, 3910 deletions
diff --git a/sound/decoders/qdm2.cpp b/sound/decoders/qdm2.cpp deleted file mode 100644 index aa4eb4b40a..0000000000 --- a/sound/decoders/qdm2.cpp +++ /dev/null @@ -1,3327 +0,0 @@ -/* ScummVM - Graphic Adventure Engine - * - * ScummVM is the legal property of its developers, whose names - * are too numerous to list here. Please refer to the COPYRIGHT - * file distributed with this source distribution. - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * as published by the Free Software Foundation; either version 2 - * of the License, or (at your option) any later version. - - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA. - * - * $URL$ - * $Id$ - * - */ - -// Based off ffmpeg's QDM2 decoder - -#include "sound/decoders/qdm2.h" - -#ifdef SOUND_QDM2_H - -#include "sound/audiostream.h" -#include "sound/decoders/qdm2data.h" - -#include "common/array.h" -#include "common/stream.h" -#include "common/system.h" - -namespace Audio { - -enum { - SOFTCLIP_THRESHOLD = 27600, - HARDCLIP_THRESHOLD = 35716, - MPA_MAX_CHANNELS = 2, - MPA_FRAME_SIZE = 1152, - FF_INPUT_BUFFER_PADDING_SIZE = 8 -}; - -typedef int8 sb_int8_array[2][30][64]; - -/* bit input */ -/* buffer, buffer_end and size_in_bits must be present and used by every reader */ -struct GetBitContext { - const uint8 *buffer, *bufferEnd; - int index; - int sizeInBits; -}; - -struct QDM2SubPacket { - int type; - unsigned int size; - const uint8 *data; // pointer to subpacket data (points to input data buffer, it's not a private copy) -}; - -struct QDM2SubPNode { - QDM2SubPacket *packet; - struct QDM2SubPNode *next; // pointer to next packet in the list, NULL if leaf node -}; - -struct QDM2Complex { - float re; - float im; -}; - -struct FFTTone { - float level; - QDM2Complex *complex; - const float *table; - int phase; - int phase_shift; - int duration; - short time_index; - short cutoff; -}; - -struct FFTCoefficient { - int16 sub_packet; - uint8 channel; - int16 offset; - int16 exp; - uint8 phase; -}; - -struct VLC { - int32 bits; - int16 (*table)[2]; // code, bits - int32 table_size; - int32 table_allocated; -}; - -#include "common/pack-start.h" -struct QDM2FFT { - QDM2Complex complex[MPA_MAX_CHANNELS][256]; -} PACKED_STRUCT; -#include "common/pack-end.h" - -enum RDFTransformType { - RDFT, - IRDFT, - RIDFT, - IRIDFT -}; - -struct FFTComplex { - float re, im; -}; - -struct FFTContext { - int nbits; - int inverse; - uint16 *revtab; - FFTComplex *exptab; - FFTComplex *tmpBuf; - int mdctSize; // size of MDCT (i.e. number of input data * 2) - int mdctBits; // n = 2^nbits - // pre/post rotation tables - float *tcos; - float *tsin; - void (*fftPermute)(struct FFTContext *s, FFTComplex *z); - void (*fftCalc)(struct FFTContext *s, FFTComplex *z); - void (*imdctCalc)(struct FFTContext *s, float *output, const float *input); - void (*imdctHalf)(struct FFTContext *s, float *output, const float *input); - void (*mdctCalc)(struct FFTContext *s, float *output, const float *input); - int splitRadix; - int permutation; -}; - -enum { - FF_MDCT_PERM_NONE = 0, - FF_MDCT_PERM_INTERLEAVE = 1 -}; - -struct RDFTContext { - int nbits; - int inverse; - int signConvention; - - // pre/post rotation tables - float *tcos; - float *tsin; - FFTContext fft; -}; - -class QDM2Stream : public Audio::AudioStream { -public: - QDM2Stream(Common::SeekableReadStream *stream, Common::SeekableReadStream *extraData); - ~QDM2Stream(); - - bool isStereo() const { return _channels == 2; } - bool endOfData() const { return ((_stream->pos() == _stream->size()) && (_outputSamples.size() == 0)); } - int getRate() const { return _sampleRate; } - int readBuffer(int16 *buffer, const int numSamples); - -private: - Common::SeekableReadStream *_stream; - - // Parameters from codec header, do not change during playback - uint8 _channels; - uint16 _sampleRate; - uint16 _bitRate; - uint16 _blockSize; // Group - uint16 _frameSize; // FFT - uint16 _packetSize; // Checksum - - // Parameters built from header parameters, do not change during playback - int _groupOrder; // order of frame group - int _fftOrder; // order of FFT (actually fft order+1) - int _fftFrameSize; // size of fft frame, in components (1 comples = re + im) - int _sFrameSize; // size of data frame - int _frequencyRange; - int _subSampling; // subsampling: 0=25%, 1=50%, 2=100% */ - int _coeffPerSbSelect; // selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2 - int _cmTableSelect; // selector for "coding method" tables. Can be 0, 1 (from init: 0-4) - - // Packets and packet lists - QDM2SubPacket _subPackets[16]; // the packets themselves - QDM2SubPNode _subPacketListA[16]; // list of all packets - QDM2SubPNode _subPacketListB[16]; // FFT packets B are on list - int _subPacketsB; // number of packets on 'B' list - QDM2SubPNode _subPacketListC[16]; // packets with errors? - QDM2SubPNode _subPacketListD[16]; // DCT packets - - // FFT and tones - FFTTone _fftTones[1000]; - int _fftToneStart; - int _fftToneEnd; - FFTCoefficient _fftCoefs[1000]; - int _fftCoefsIndex; - int _fftCoefsMinIndex[5]; - int _fftCoefsMaxIndex[5]; - int _fftLevelExp[6]; - //RDFTContext _rdftCtx; - QDM2FFT _fft; - - // I/O data - uint8 *_compressedData; - float _outputBuffer[1024]; - Common::Array<int16> _outputSamples; - - // Synthesis filter - int16 ff_mpa_synth_window[512]; - int16 _synthBuf[MPA_MAX_CHANNELS][512*2]; - int _synthBufOffset[MPA_MAX_CHANNELS]; - int32 _sbSamples[MPA_MAX_CHANNELS][128][32]; - - // Mixed temporary data used in decoding - float _toneLevel[MPA_MAX_CHANNELS][30][64]; - int8 _codingMethod[MPA_MAX_CHANNELS][30][64]; - int8 _quantizedCoeffs[MPA_MAX_CHANNELS][10][8]; - int8 _toneLevelIdxBase[MPA_MAX_CHANNELS][30][8]; - int8 _toneLevelIdxHi1[MPA_MAX_CHANNELS][3][8][8]; - int8 _toneLevelIdxMid[MPA_MAX_CHANNELS][26][8]; - int8 _toneLevelIdxHi2[MPA_MAX_CHANNELS][26]; - int8 _toneLevelIdx[MPA_MAX_CHANNELS][30][64]; - int8 _toneLevelIdxTemp[MPA_MAX_CHANNELS][30][64]; - - // Flags - bool _hasErrors; // packet has errors - int _superblocktype_2_3; // select fft tables and some algorithm based on superblock type - int _doSynthFilter; // used to perform or skip synthesis filter - - uint8 _subPacket; // 0 to 15 - int _noiseIdx; // index for dithering noise table - - byte _emptyBuffer[FF_INPUT_BUFFER_PADDING_SIZE]; - - VLC _vlcTabLevel; - VLC _vlcTabDiff; - VLC _vlcTabRun; - VLC _fftLevelExpAltVlc; - VLC _fftLevelExpVlc; - VLC _fftStereoExpVlc; - VLC _fftStereoPhaseVlc; - VLC _vlcTabToneLevelIdxHi1; - VLC _vlcTabToneLevelIdxMid; - VLC _vlcTabToneLevelIdxHi2; - VLC _vlcTabType30; - VLC _vlcTabType34; - VLC _vlcTabFftToneOffset[5]; - bool _vlcsInitialized; - void initVlc(void); - - uint16 _softclipTable[HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1]; - void softclipTableInit(void); - - float _noiseTable[4096]; - byte _randomDequantIndex[256][5]; - byte _randomDequantType24[128][3]; - void rndTableInit(void); - - float _noiseSamples[128]; - void initNoiseSamples(void); - - RDFTContext _rdftCtx; - - void average_quantized_coeffs(void); - void build_sb_samples_from_noise(int sb); - void fix_coding_method_array(int sb, int channels, sb_int8_array coding_method); - void fill_tone_level_array(int flag); - void fill_coding_method_array(sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp, - sb_int8_array coding_method, int nb_channels, - int c, int superblocktype_2_3, int cm_table_select); - void synthfilt_build_sb_samples(GetBitContext *gb, int length, int sb_min, int sb_max); - void init_quantized_coeffs_elem0(int8 *quantized_coeffs, GetBitContext *gb, int length); - void init_tone_level_dequantization(GetBitContext *gb, int length); - void process_subpacket_9(QDM2SubPNode *node); - void process_subpacket_10(QDM2SubPNode *node, int length); - void process_subpacket_11(QDM2SubPNode *node, int length); - void process_subpacket_12(QDM2SubPNode *node, int length); - void process_synthesis_subpackets(QDM2SubPNode *list); - void qdm2_decode_super_block(void); - void qdm2_fft_init_coefficient(int sub_packet, int offset, int duration, - int channel, int exp, int phase); - void qdm2_fft_decode_tones(int duration, GetBitContext *gb, int b); - void qdm2_decode_fft_packets(void); - void qdm2_fft_generate_tone(FFTTone *tone); - void qdm2_fft_tone_synthesizer(uint8 sub_packet); - void qdm2_calculate_fft(int channel); - void qdm2_synthesis_filter(uint8 index); - int qdm2_decodeFrame(Common::SeekableReadStream *in); -}; - -// Fix compilation for non C99-compliant compilers, like MSVC -#ifndef int64_t -typedef signed long long int int64_t; -#endif - -// Integer log2 function. This is much faster than invoking -// double precision C99 log2 math functions or equivalent, since -// this is only used to determine maximum number of bits needed -// i.e. only non-fractional part is needed. Also, the double -// version is incorrect for exact cases due to floating point -// rounding errors. -static inline int scummvm_log2(int n) { - int ret = -1; - while(n != 0) { - n /= 2; - ret++; - } - return ret; -} - -#define QDM2_LIST_ADD(list, size, packet) \ - do { \ - if (size > 0) \ - list[size - 1].next = &list[size]; \ - list[size].packet = packet; \ - list[size].next = NULL; \ - size++; \ - } while(0) - -// Result is 8, 16 or 30 -#define QDM2_SB_USED(subSampling) (((subSampling) >= 2) ? 30 : 8 << (subSampling)) - -#define FIX_NOISE_IDX(noiseIdx) \ - if ((noiseIdx) >= 3840) \ - (noiseIdx) -= 3840 \ - -#define SB_DITHERING_NOISE(sb, noiseIdx) (_noiseTable[(noiseIdx)++] * sb_noise_attenuation[(sb)]) - -static inline void initGetBits(GetBitContext *s, const uint8 *buffer, int bitSize) { - int bufferSize = (bitSize + 7) >> 3; - - debug(1, "void initGetBits(GetBitContext *s, const uint8 *buffer, int bitSize)"); - - if (bufferSize < 0 || bitSize < 0) { - bufferSize = bitSize = 0; - buffer = NULL; - } - - s->buffer = buffer; - s->sizeInBits = bitSize; - s->bufferEnd = buffer + bufferSize; - s->index = 0; -} - -static inline int getBitsCount(GetBitContext *s) { - debug(1, "int getBitsCount(GetBitContext *s)"); - return s->index; -} - -static inline unsigned int getBits1(GetBitContext *s) { - int index; - uint8 result; - - debug(1, "unsigned int getBits1(GetBitContext *s)"); - - index = s->index; - result = s->buffer[index >> 3]; - - debug(1, "index : %d", index); - - result >>= (index & 0x07); - result &= 1; - index++; - s->index = index; - - return result; -} - -static inline unsigned int getBits(GetBitContext *s, int n) { - int tmp, reCache, reIndex; - - debug(1, "unsigned int getBits(GetBitContext *s, int n)"); - - reIndex = s->index; - - debug(1, "reIndex : %d", reIndex); - - reCache = READ_LE_UINT32((const uint8 *)s->buffer + (reIndex >> 3)) >> (reIndex & 0x07); - - tmp = (reCache) & ((uint32)0xffffffff >> (32 - n)); - - s->index = reIndex + n; - - return tmp; -} - -static inline void skipBits(GetBitContext *s, int n) { - int reIndex, reCache; - - debug(1, "void skipBits(GetBitContext *s, int n)"); - - reIndex = s->index; - reCache = 0; - - debug(1, "reIndex : %d", reIndex); - - reCache = READ_LE_UINT32((const uint8 *)s->buffer + (reIndex >> 3)) >> (reIndex & 0x07); - s->index = reIndex + n; -} - -#define BITS_LEFT(length, gb) ((length) - getBitsCount((gb))) - -static int splitRadixPermutation(int i, int n, int inverse) { - if (n <= 2) - return i & 1; - - int m = n >> 1; - - if(!(i & m)) - return splitRadixPermutation(i, m, inverse) * 2; - - m >>= 1; - - if (inverse == !(i & m)) - return splitRadixPermutation(i, m, inverse) * 4 + 1; - - return splitRadixPermutation(i, m, inverse) * 4 - 1; -} - -// sin(2*pi*x/n) for 0<=x<n/4, followed by n/2<=x<3n/4 -float ff_sin_16[8]; -float ff_sin_32[16]; -float ff_sin_64[32]; -float ff_sin_128[64]; -float ff_sin_256[128]; -float ff_sin_512[256]; -float ff_sin_1024[512]; -float ff_sin_2048[1024]; -float ff_sin_4096[2048]; -float ff_sin_8192[4096]; -float ff_sin_16384[8192]; -float ff_sin_32768[16384]; -float ff_sin_65536[32768]; - -float *ff_sin_tabs[] = { - NULL, NULL, NULL, NULL, - ff_sin_16, ff_sin_32, ff_sin_64, ff_sin_128, ff_sin_256, ff_sin_512, ff_sin_1024, - ff_sin_2048, ff_sin_4096, ff_sin_8192, ff_sin_16384, ff_sin_32768, ff_sin_65536, -}; - -// cos(2*pi*x/n) for 0<=x<=n/4, followed by its reverse -float ff_cos_16[8]; -float ff_cos_32[16]; -float ff_cos_64[32]; -float ff_cos_128[64]; -float ff_cos_256[128]; -float ff_cos_512[256]; -float ff_cos_1024[512]; -float ff_cos_2048[1024]; -float ff_cos_4096[2048]; -float ff_cos_8192[4096]; -float ff_cos_16384[8192]; -float ff_cos_32768[16384]; -float ff_cos_65536[32768]; - -float *ff_cos_tabs[] = { - NULL, NULL, NULL, NULL, - ff_cos_16, ff_cos_32, ff_cos_64, ff_cos_128, ff_cos_256, ff_cos_512, ff_cos_1024, - ff_cos_2048, ff_cos_4096, ff_cos_8192, ff_cos_16384, ff_cos_32768, ff_cos_65536, -}; - -void initCosineTables(int index) { - int m = 1 << index; - double freq = 2 * PI / m; - float *tab = ff_cos_tabs[index]; - - for (int i = 0; i <= m / 4; i++) - tab[i] = cos(i * freq); - - for (int i = 1; i < m / 4; i++) - tab[m / 2 - i] = tab[i]; -} - -void fftPermute(FFTContext *s, FFTComplex *z) { - const uint16 *revtab = s->revtab; - int np = 1 << s->nbits; - - if (s->tmpBuf) { - // TODO: handle split-radix permute in a more optimal way, probably in-place - for (int j = 0; j < np; j++) - s->tmpBuf[revtab[j]] = z[j]; - memcpy(z, s->tmpBuf, np * sizeof(FFTComplex)); - return; - } - - // reverse - for (int j = 0; j < np; j++) { - int k = revtab[j]; - if (k < j) { - FFTComplex tmp = z[k]; - z[k] = z[j]; - z[j] = tmp; - } - } -} - -#define DECL_FFT(n,n2,n4) \ -static void fft##n(FFTComplex *z) { \ - fft##n2(z); \ - fft##n4(z + n4 * 2); \ - fft##n4(z + n4 * 3); \ - pass(z, ff_cos_##n, n4 / 2); \ -} - -#ifndef M_SQRT1_2 -#define M_SQRT1_2 7.0710678118654752440E-1 -#endif - -#define sqrthalf (float)M_SQRT1_2 - -#define BF(x,y,a,b) { \ - x = a - b; \ - y = a + b; \ -} - -#define BUTTERFLIES(a0, a1, a2, a3) { \ - BF(t3, t5, t5, t1); \ - BF(a2.re, a0.re, a0.re, t5); \ - BF(a3.im, a1.im, a1.im, t3); \ - BF(t4, t6, t2, t6); \ - BF(a3.re, a1.re, a1.re, t4); \ - BF(a2.im, a0.im, a0.im, t6); \ -} - -// force loading all the inputs before storing any. -// this is slightly slower for small data, but avoids store->load aliasing -// for addresses separated by large powers of 2. -#define BUTTERFLIES_BIG(a0, a1, a2, a3) { \ - float r0 = a0.re, i0 = a0.im, r1 = a1.re, i1 = a1.im; \ - BF(t3, t5, t5, t1); \ - BF(a2.re, a0.re, r0, t5); \ - BF(a3.im, a1.im, i1, t3); \ - BF(t4, t6, t2, t6); \ - BF(a3.re, a1.re, r1, t4); \ - BF(a2.im, a0.im, i0, t6); \ -} - -#define TRANSFORM(a0, a1, a2, a3, wre, wim) { \ - t1 = a2.re * wre + a2.im * wim; \ - t2 = a2.im * wre - a2.re * wim; \ - t5 = a3.re * wre - a3.im * wim; \ - t6 = a3.im * wre + a3.re * wim; \ - BUTTERFLIES(a0, a1, a2, a3) \ -} - -#define TRANSFORM_ZERO(a0, a1, a2, a3) { \ - t1 = a2.re; \ - t2 = a2.im; \ - t5 = a3.re; \ - t6 = a3.im; \ - BUTTERFLIES(a0, a1, a2, a3) \ -} - -// z[0...8n-1], w[1...2n-1] -#define PASS(name) \ -static void name(FFTComplex *z, const float *wre, unsigned int n) { \ - float t1, t2, t3, t4, t5, t6; \ - int o1 = 2 * n; \ - int o2 = 4 * n; \ - int o3 = 6 * n; \ - const float *wim = wre + o1; \ - n--; \ - \ - TRANSFORM_ZERO(z[0], z[o1], z[o2], z[o3]); \ - TRANSFORM(z[1], z[o1 + 1], z[o2 + 1], z[o3 + 1], wre[1], wim[-1]); \ - \ - do { \ - z += 2; \ - wre += 2; \ - wim -= 2; \ - TRANSFORM(z[0], z[o1], z[o2], z[o3], wre[0], wim[0]); \ - TRANSFORM(z[1], z[o1 + 1],z[o2 + 1], z[o3 + 1], wre[1], wim[-1]); \ - } while(--n); \ -} - -PASS(pass) -#undef BUTTERFLIES -#define BUTTERFLIES BUTTERFLIES_BIG -PASS(pass_big) - -static void fft4(FFTComplex *z) { - float t1, t2, t3, t4, t5, t6, t7, t8; - - BF(t3, t1, z[0].re, z[1].re); - BF(t8, t6, z[3].re, z[2].re); - BF(z[2].re, z[0].re, t1, t6); - BF(t4, t2, z[0].im, z[1].im); - BF(t7, t5, z[2].im, z[3].im); - BF(z[3].im, z[1].im, t4, t8); - BF(z[3].re, z[1].re, t3, t7); - BF(z[2].im, z[0].im, t2, t5); -} - -static void fft8(FFTComplex *z) { - float t1, t2, t3, t4, t5, t6, t7, t8; - - fft4(z); - - BF(t1, z[5].re, z[4].re, -z[5].re); - BF(t2, z[5].im, z[4].im, -z[5].im); - BF(t3, z[7].re, z[6].re, -z[7].re); - BF(t4, z[7].im, z[6].im, -z[7].im); - BF(t8, t1, t3, t1); - BF(t7, t2, t2, t4); - BF(z[4].re, z[0].re, z[0].re, t1); - BF(z[4].im, z[0].im, z[0].im, t2); - BF(z[6].re, z[2].re, z[2].re, t7); - BF(z[6].im, z[2].im, z[2].im, t8); - - TRANSFORM(z[1], z[3], z[5], z[7], sqrthalf, sqrthalf); -} - -#undef BF - -DECL_FFT(16,8,4) -DECL_FFT(32,16,8) -DECL_FFT(64,32,16) -DECL_FFT(128,64,32) -DECL_FFT(256,128,64) -DECL_FFT(512,256,128) -#define pass pass_big -DECL_FFT(1024,512,256) -DECL_FFT(2048,1024,512) -DECL_FFT(4096,2048,1024) -DECL_FFT(8192,4096,2048) -DECL_FFT(16384,8192,4096) -DECL_FFT(32768,16384,8192) -DECL_FFT(65536,32768,16384) - -void fftCalc(FFTContext *s, FFTComplex *z) { - static void (* const fftDispatch[])(FFTComplex*) = { - fft4, fft8, fft16, fft32, fft64, fft128, fft256, fft512, fft1024, - fft2048, fft4096, fft8192, fft16384, fft32768, fft65536, - }; - - fftDispatch[s->nbits - 2](z); -} - -// complex multiplication: p = a * b -#define CMUL(pre, pim, are, aim, bre, bim) \ -{\ - float _are = (are); \ - float _aim = (aim); \ - float _bre = (bre); \ - float _bim = (bim); \ - (pre) = _are * _bre - _aim * _bim; \ - (pim) = _are * _bim + _aim * _bre; \ -} - -/** - * Compute the middle half of the inverse MDCT of size N = 2^nbits, - * thus excluding the parts that can be derived by symmetry - * @param output N/2 samples - * @param input N/2 samples - */ -void imdctHalfC(FFTContext *s, float *output, const float *input) { - const uint16 *revtab = s->revtab; - const float *tcos = s->tcos; - const float *tsin = s->tsin; - FFTComplex *z = (FFTComplex *)output; - - int n = 1 << s->mdctBits; - int n2 = n >> 1; - int n4 = n >> 2; - int n8 = n >> 3; - - // pre rotation - const float *in1 = input; - const float *in2 = input + n2 - 1; - for (int k = 0; k < n4; k++) { - int j = revtab[k]; - CMUL(z[j].re, z[j].im, *in2, *in1, tcos[k], tsin[k]); - in1 += 2; - in2 -= 2; - } - - fftCalc(s, z); - - // post rotation + reordering - for (int k = 0; k < n8; k++) { - float r0, i0, r1, i1; - CMUL(r0, i1, z[n8 - k - 1].im, z[n8 - k - 1].re, tsin[n8 - k - 1], tcos[n8 - k - 1]); - CMUL(r1, i0, z[n8 + k].im, z[n8 + k].re, tsin[n8 + k], tcos[n8 + k]); - z[n8 - k - 1].re = r0; - z[n8 - k - 1].im = i0; - z[n8 + k].re = r1; - z[n8 + k].im = i1; - } -} - -/** - * Compute inverse MDCT of size N = 2^nbits - * @param output N samples - * @param input N/2 samples - */ -void imdctCalcC(FFTContext *s, float *output, const float *input) { - int n = 1 << s->mdctBits; - int n2 = n >> 1; - int n4 = n >> 2; - - imdctHalfC(s, output + n4, input); - - for (int k = 0; k < n4; k++) { - output[k] = -output[n2 - k - 1]; - output[n - k - 1] = output[n2 + k]; - } -} - -/** - * Compute MDCT of size N = 2^nbits - * @param input N samples - * @param out N/2 samples - */ -void mdctCalcC(FFTContext *s, float *out, const float *input) { - const uint16 *revtab = s->revtab; - const float *tcos = s->tcos; - const float *tsin = s->tsin; - FFTComplex *x = (FFTComplex *)out; - - int n = 1 << s->mdctBits; - int n2 = n >> 1; - int n4 = n >> 2; - int n8 = n >> 3; - int n3 = 3 * n4; - - // pre rotation - for (int i = 0; i < n8; i++) { - float re = -input[2 * i + 3 * n4] - input[n3 - 1 - 2 * i]; - float im = -input[n4 + 2 * i] + input[n4 - 1 - 2 * i]; - int j = revtab[i]; - CMUL(x[j].re, x[j].im, re, im, -tcos[i], tsin[i]); - - re = input[2 * i] - input[n2 - 1 - 2 * i]; - im = -(input[n2 + 2 * i] + input[n - 1 - 2 * i]); - j = revtab[n8 + i]; - CMUL(x[j].re, x[j].im, re, im, -tcos[n8 + i], tsin[n8 + i]); - } - - fftCalc(s, x); - - // post rotation - for (int i = 0; i < n8; i++) { - float r0, i0, r1, i1; - CMUL(i1, r0, x[n8 - i - 1].re, x[n8 - i - 1].im, -tsin[n8 - i - 1], -tcos[n8 - i - 1]); - CMUL(i0, r1, x[n8 + i].re, x[n8 + i].im, -tsin[n8 + i], -tcos[n8 + i]); - x[n8 - i - 1].re = r0; - x[n8 - i - 1].im = i0; - x[n8 + i].re = r1; - x[n8 + i].im = i1; - } -} - -int fftInit(FFTContext *s, int nbits, int inverse) { - int i, j, m, n; - float alpha, c1, s1, s2; - - if (nbits < 2 || nbits > 16) - goto fail; - - s->nbits = nbits; - n = 1 << nbits; - s->tmpBuf = NULL; - - s->exptab = (FFTComplex *)malloc((n / 2) * sizeof(FFTComplex)); - if (!s->exptab) - goto fail; - - s->revtab = (uint16 *)malloc(n * sizeof(uint16)); - if (!s->revtab) - goto fail; - s->inverse = inverse; - - s2 = inverse ? 1.0 : -1.0; - - s->fftPermute = fftPermute; - s->fftCalc = fftCalc; - s->imdctCalc = imdctCalcC; - s->imdctHalf = imdctHalfC; - s->mdctCalc = mdctCalcC; - s->splitRadix = 1; - - if (s->splitRadix) { - for (j = 4; j <= nbits; j++) - initCosineTables(j); - - for (i = 0; i < n; i++) - s->revtab[-splitRadixPermutation(i, n, s->inverse) & (n - 1)] = i; - - s->tmpBuf = (FFTComplex *)malloc(n * sizeof(FFTComplex)); - } else { - for (i = 0; i < n / 2; i++) { - alpha = 2 * PI * (float)i / (float)n; - c1 = cos(alpha); - s1 = sin(alpha) * s2; - s->exptab[i].re = c1; - s->exptab[i].im = s1; - } - - //int np = 1 << nbits; - //int nblocks = np >> 3; - //int np2 = np >> 1; - - // compute bit reverse table - for (i = 0; i < n; i++) { - m = 0; - - for (j = 0; j < nbits; j++) - m |= ((i >> j) & 1) << (nbits - j - 1); - - s->revtab[i] = m; - } - } - - return 0; - - fail: - free(&s->revtab); - free(&s->exptab); - free(&s->tmpBuf); - return -1; -} - -/** - * Sets up a real FFT. - * @param nbits log2 of the length of the input array - * @param trans the type of transform - */ -int rdftInit(RDFTContext *s, int nbits, RDFTransformType trans) { - int n = 1 << nbits; - const double theta = (trans == RDFT || trans == IRIDFT ? -1 : 1) * 2 * PI / n; - - s->nbits = nbits; - s->inverse = trans == IRDFT || trans == IRIDFT; - s->signConvention = trans == RIDFT || trans == IRIDFT ? 1 : -1; - - if (nbits < 4 || nbits > 16) - return -1; - - if (fftInit(&s->fft, nbits - 1, trans == IRDFT || trans == RIDFT) < 0) - return -1; - - initCosineTables(nbits); - s->tcos = ff_cos_tabs[nbits]; - s->tsin = ff_sin_tabs[nbits] + (trans == RDFT || trans == IRIDFT) * (n >> 2); - - for (int i = 0; i < n >> 2; i++) - s->tsin[i] = sin(i*theta); - - return 0; -} - -/** Map one real FFT into two parallel real even and odd FFTs. Then interleave - * the two real FFTs into one complex FFT. Unmangle the results. - * ref: http://www.engineeringproductivitytools.com/stuff/T0001/PT10.HTM - */ -void rdftCalc(RDFTContext *s, float *data) { - FFTComplex ev, od; - - const int n = 1 << s->nbits; - const float k1 = 0.5; - const float k2 = 0.5 - s->inverse; - const float *tcos = s->tcos; - const float *tsin = s->tsin; - - if (!s->inverse) { - fftPermute(&s->fft, (FFTComplex *)data); - fftCalc(&s->fft, (FFTComplex *)data); - } - - // i=0 is a special case because of packing, the DC term is real, so we - // are going to throw the N/2 term (also real) in with it. - ev.re = data[0]; - data[0] = ev.re + data[1]; - data[1] = ev.re - data[1]; - - int i; - - for (i = 1; i < n >> 2; i++) { - int i1 = i * 2; - int i2 = n - i1; - - // Separate even and odd FFTs - ev.re = k1 * (data[i1] + data[i2]); - od.im = -k2 * (data[i1] - data[i2]); - ev.im = k1 * (data[i1 + 1] - data[i2 + 1]); - od.re = k2 * (data[i1 + 1] + data[i2 + 1]); - - // Apply twiddle factors to the odd FFT and add to the even FFT - data[i1] = ev.re + od.re * tcos[i] - od.im * tsin[i]; - data[i1 + 1] = ev.im + od.im * tcos[i] + od.re * tsin[i]; - data[i2] = ev.re - od.re * tcos[i] + od.im * tsin[i]; - data[i2 + 1] = -ev.im + od.im * tcos[i] + od.re * tsin[i]; - } - - data[i * 2 + 1] = s->signConvention * data[i * 2 + 1]; - if (s->inverse) { - data[0] *= k1; - data[1] *= k1; - fftPermute(&s->fft, (FFTComplex*)data); - fftCalc(&s->fft, (FFTComplex*)data); - } -} - -// half mpeg encoding window (full precision) -const int32 ff_mpa_enwindow[257] = { - 0, -1, -1, -1, -1, -1, -1, -2, - -2, -2, -2, -3, -3, -4, -4, -5, - -5, -6, -7, -7, -8, -9, -10, -11, - -13, -14, -16, -17, -19, -21, -24, -26, - -29, -31, -35, -38, -41, -45, -49, -53, - -58, -63, -68, -73, -79, -85, -91, -97, - -104, -111, -117, -125, -132, -139, -147, -154, - -161, -169, -176, -183, -190, -196, -202, -208, - 213, 218, 222, 225, 227, 228, 228, 227, - 224, 221, 215, 208, 200, 189, 177, 163, - 146, 127, 106, 83, 57, 29, -2, -36, - -72, -111, -153, -197, -244, -294, -347, -401, - -459, -519, -581, -645, -711, -779, -848, -919, - -991, -1064, -1137, -1210, -1283, -1356, -1428, -1498, - -1567, -1634, -1698, -1759, -1817, -1870, -1919, -1962, - -2001, -2032, -2057, -2075, -2085, -2087, -2080, -2063, - 2037, 2000, 1952, 1893, 1822, 1739, 1644, 1535, - 1414, 1280, 1131, 970, 794, 605, 402, 185, - -45, -288, -545, -814, -1095, -1388, -1692, -2006, - -2330, -2663, -3004, -3351, -3705, -4063, -4425, -4788, - -5153, -5517, -5879, -6237, -6589, -6935, -7271, -7597, - -7910, -8209, -8491, -8755, -8998, -9219, -9416, -9585, - -9727, -9838, -9916, -9959, -9966, -9935, -9863, -9750, - -9592, -9389, -9139, -8840, -8492, -8092, -7640, -7134, - 6574, 5959, 5288, 4561, 3776, 2935, 2037, 1082, - 70, -998, -2122, -3300, -4533, -5818, -7154, -8540, - -9975,-11455,-12980,-14548,-16155,-17799,-19478,-21189, --22929,-24694,-26482,-28289,-30112,-31947,-33791,-35640, --37489,-39336,-41176,-43006,-44821,-46617,-48390,-50137, --51853,-53534,-55178,-56778,-58333,-59838,-61289,-62684, --64019,-65290,-66494,-67629,-68692,-69679,-70590,-71420, --72169,-72835,-73415,-73908,-74313,-74630,-74856,-74992, - 75038 -}; - -void ff_mpa_synth_init(int16 *window) { - int i; - int32 v; - - // max = 18760, max sum over all 16 coefs : 44736 - for(i = 0; i < 257; i++) { - v = ff_mpa_enwindow[i]; - v = (v + 2) >> 2; - window[i] = v; - - if ((i & 63) != 0) - v = -v; - - if (i != 0) - window[512 - i] = v; - } -} - -static inline uint16 round_sample(int *sum) { - int sum1; - sum1 = (*sum) >> 14; - *sum &= (1 << 14)-1; - if (sum1 < (-0x7fff - 1)) - sum1 = (-0x7fff - 1); - if (sum1 > 0x7fff) - sum1 = 0x7fff; - return sum1; -} - -static inline int MULH(int a, int b) { - return ((int64_t)(a) * (int64_t)(b))>>32; -} - -// signed 16x16 -> 32 multiply add accumulate -#define MACS(rt, ra, rb) rt += (ra) * (rb) - -#define MLSS(rt, ra, rb) ((rt) -= (ra) * (rb)) - -#define SUM8(op, sum, w, p)\ -{\ - op(sum, (w)[0 * 64], (p)[0 * 64]);\ - op(sum, (w)[1 * 64], (p)[1 * 64]);\ - op(sum, (w)[2 * 64], (p)[2 * 64]);\ - op(sum, (w)[3 * 64], (p)[3 * 64]);\ - op(sum, (w)[4 * 64], (p)[4 * 64]);\ - op(sum, (w)[5 * 64], (p)[5 * 64]);\ - op(sum, (w)[6 * 64], (p)[6 * 64]);\ - op(sum, (w)[7 * 64], (p)[7 * 64]);\ -} - -#define SUM8P2(sum1, op1, sum2, op2, w1, w2, p) \ -{\ - tmp_s = p[0 * 64];\ - op1(sum1, (w1)[0 * 64], tmp_s);\ - op2(sum2, (w2)[0 * 64], tmp_s);\ - tmp_s = p[1 * 64];\ - op1(sum1, (w1)[1 * 64], tmp_s);\ - op2(sum2, (w2)[1 * 64], tmp_s);\ - tmp_s = p[2 * 64];\ - op1(sum1, (w1)[2 * 64], tmp_s);\ - op2(sum2, (w2)[2 * 64], tmp_s);\ - tmp_s = p[3 * 64];\ - op1(sum1, (w1)[3 * 64], tmp_s);\ - op2(sum2, (w2)[3 * 64], tmp_s);\ - tmp_s = p[4 * 64];\ - op1(sum1, (w1)[4 * 64], tmp_s);\ - op2(sum2, (w2)[4 * 64], tmp_s);\ - tmp_s = p[5 * 64];\ - op1(sum1, (w1)[5 * 64], tmp_s);\ - op2(sum2, (w2)[5 * 64], tmp_s);\ - tmp_s = p[6 * 64];\ - op1(sum1, (w1)[6 * 64], tmp_s);\ - op2(sum2, (w2)[6 * 64], tmp_s);\ - tmp_s = p[7 * 64];\ - op1(sum1, (w1)[7 * 64], tmp_s);\ - op2(sum2, (w2)[7 * 64], tmp_s);\ -} - -#define FIXHR(a) ((int)((a) * (1LL<<32) + 0.5)) - -// tab[i][j] = 1.0 / (2.0 * cos(pi*(2*k+1) / 2^(6 - j))) - -// cos(i*pi/64) - -#define COS0_0 FIXHR(0.50060299823519630134/2) -#define COS0_1 FIXHR(0.50547095989754365998/2) -#define COS0_2 FIXHR(0.51544730992262454697/2) -#define COS0_3 FIXHR(0.53104259108978417447/2) -#define COS0_4 FIXHR(0.55310389603444452782/2) -#define COS0_5 FIXHR(0.58293496820613387367/2) -#define COS0_6 FIXHR(0.62250412303566481615/2) -#define COS0_7 FIXHR(0.67480834145500574602/2) -#define COS0_8 FIXHR(0.74453627100229844977/2) -#define COS0_9 FIXHR(0.83934964541552703873/2) -#define COS0_10 FIXHR(0.97256823786196069369/2) -#define COS0_11 FIXHR(1.16943993343288495515/4) -#define COS0_12 FIXHR(1.48416461631416627724/4) -#define COS0_13 FIXHR(2.05778100995341155085/8) -#define COS0_14 FIXHR(3.40760841846871878570/8) -#define COS0_15 FIXHR(10.19000812354805681150/32) - -#define COS1_0 FIXHR(0.50241928618815570551/2) -#define COS1_1 FIXHR(0.52249861493968888062/2) -#define COS1_2 FIXHR(0.56694403481635770368/2) -#define COS1_3 FIXHR(0.64682178335999012954/2) -#define COS1_4 FIXHR(0.78815462345125022473/2) -#define COS1_5 FIXHR(1.06067768599034747134/4) -#define COS1_6 FIXHR(1.72244709823833392782/4) -#define COS1_7 FIXHR(5.10114861868916385802/16) - -#define COS2_0 FIXHR(0.50979557910415916894/2) -#define COS2_1 FIXHR(0.60134488693504528054/2) -#define COS2_2 FIXHR(0.89997622313641570463/2) -#define COS2_3 FIXHR(2.56291544774150617881/8) - -#define COS3_0 FIXHR(0.54119610014619698439/2) -#define COS3_1 FIXHR(1.30656296487637652785/4) - -#define COS4_0 FIXHR(0.70710678118654752439/2) - -/* butterfly operator */ -#define BF(a, b, c, s)\ -{\ - tmp0 = tab[a] + tab[b];\ - tmp1 = tab[a] - tab[b];\ - tab[a] = tmp0;\ - tab[b] = MULH(tmp1<<(s), c);\ -} - -#define BF1(a, b, c, d)\ -{\ - BF(a, b, COS4_0, 1);\ - BF(c, d,-COS4_0, 1);\ - tab[c] += tab[d];\ -} - -#define BF2(a, b, c, d)\ -{\ - BF(a, b, COS4_0, 1);\ - BF(c, d,-COS4_0, 1);\ - tab[c] += tab[d];\ - tab[a] += tab[c];\ - tab[c] += tab[b];\ - tab[b] += tab[d];\ -} - -#define ADD(a, b) tab[a] += tab[b] - -// DCT32 without 1/sqrt(2) coef zero scaling. -static void dct32(int32 *out, int32 *tab) { - int tmp0, tmp1; - - // pass 1 - BF( 0, 31, COS0_0 , 1); - BF(15, 16, COS0_15, 5); - // pass 2 - BF( 0, 15, COS1_0 , 1); - BF(16, 31,-COS1_0 , 1); - // pass 1 - BF( 7, 24, COS0_7 , 1); - BF( 8, 23, COS0_8 , 1); - // pass 2 - BF( 7, 8, COS1_7 , 4); - BF(23, 24,-COS1_7 , 4); - // pass 3 - BF( 0, 7, COS2_0 , 1); - BF( 8, 15,-COS2_0 , 1); - BF(16, 23, COS2_0 , 1); - BF(24, 31,-COS2_0 , 1); - // pass 1 - BF( 3, 28, COS0_3 , 1); - BF(12, 19, COS0_12, 2); - // pass 2 - BF( 3, 12, COS1_3 , 1); - BF(19, 28,-COS1_3 , 1); - // pass 1 - BF( 4, 27, COS0_4 , 1); - BF(11, 20, COS0_11, 2); - // pass 2 - BF( 4, 11, COS1_4 , 1); - BF(20, 27,-COS1_4 , 1); - // pass 3 - BF( 3, 4, COS2_3 , 3); - BF(11, 12,-COS2_3 , 3); - BF(19, 20, COS2_3 , 3); - BF(27, 28,-COS2_3 , 3); - // pass 4 - BF( 0, 3, COS3_0 , 1); - BF( 4, 7,-COS3_0 , 1); - BF( 8, 11, COS3_0 , 1); - BF(12, 15,-COS3_0 , 1); - BF(16, 19, COS3_0 , 1); - BF(20, 23,-COS3_0 , 1); - BF(24, 27, COS3_0 , 1); - BF(28, 31,-COS3_0 , 1); - - // pass 1 - BF( 1, 30, COS0_1 , 1); - BF(14, 17, COS0_14, 3); - // pass 2 - BF( 1, 14, COS1_1 , 1); - BF(17, 30,-COS1_1 , 1); - // pass 1 - BF( 6, 25, COS0_6 , 1); - BF( 9, 22, COS0_9 , 1); - // pass 2 - BF( 6, 9, COS1_6 , 2); - BF(22, 25,-COS1_6 , 2); - // pass 3 - BF( 1, 6, COS2_1 , 1); - BF( 9, 14,-COS2_1 , 1); - BF(17, 22, COS2_1 , 1); - BF(25, 30,-COS2_1 , 1); - - // pass 1 - BF( 2, 29, COS0_2 , 1); - BF(13, 18, COS0_13, 3); - // pass 2 - BF( 2, 13, COS1_2 , 1); - BF(18, 29,-COS1_2 , 1); - // pass 1 - BF( 5, 26, COS0_5 , 1); - BF(10, 21, COS0_10, 1); - // pass 2 - BF( 5, 10, COS1_5 , 2); - BF(21, 26,-COS1_5 , 2); - // pass 3 - BF( 2, 5, COS2_2 , 1); - BF(10, 13,-COS2_2 , 1); - BF(18, 21, COS2_2 , 1); - BF(26, 29,-COS2_2 , 1); - // pass 4 - BF( 1, 2, COS3_1 , 2); - BF( 5, 6,-COS3_1 , 2); - BF( 9, 10, COS3_1 , 2); - BF(13, 14,-COS3_1 , 2); - BF(17, 18, COS3_1 , 2); - BF(21, 22,-COS3_1 , 2); - BF(25, 26, COS3_1 , 2); - BF(29, 30,-COS3_1 , 2); - - // pass 5 - BF1( 0, 1, 2, 3); - BF2( 4, 5, 6, 7); - BF1( 8, 9, 10, 11); - BF2(12, 13, 14, 15); - BF1(16, 17, 18, 19); - BF2(20, 21, 22, 23); - BF1(24, 25, 26, 27); - BF2(28, 29, 30, 31); - - // pass 6 - ADD( 8, 12); - ADD(12, 10); - ADD(10, 14); - ADD(14, 9); - ADD( 9, 13); - ADD(13, 11); - ADD(11, 15); - - out[ 0] = tab[0]; - out[16] = tab[1]; - out[ 8] = tab[2]; - out[24] = tab[3]; - out[ 4] = tab[4]; - out[20] = tab[5]; - out[12] = tab[6]; - out[28] = tab[7]; - out[ 2] = tab[8]; - out[18] = tab[9]; - out[10] = tab[10]; - out[26] = tab[11]; - out[ 6] = tab[12]; - out[22] = tab[13]; - out[14] = tab[14]; - out[30] = tab[15]; - - ADD(24, 28); - ADD(28, 26); - ADD(26, 30); - ADD(30, 25); - ADD(25, 29); - ADD(29, 27); - ADD(27, 31); - - out[ 1] = tab[16] + tab[24]; - out[17] = tab[17] + tab[25]; - out[ 9] = tab[18] + tab[26]; - out[25] = tab[19] + tab[27]; - out[ 5] = tab[20] + tab[28]; - out[21] = tab[21] + tab[29]; - out[13] = tab[22] + tab[30]; - out[29] = tab[23] + tab[31]; - out[ 3] = tab[24] + tab[20]; - out[19] = tab[25] + tab[21]; - out[11] = tab[26] + tab[22]; - out[27] = tab[27] + tab[23]; - out[ 7] = tab[28] + tab[18]; - out[23] = tab[29] + tab[19]; - out[15] = tab[30] + tab[17]; - out[31] = tab[31]; -} - -// 32 sub band synthesis filter. Input: 32 sub band samples, Output: -// 32 samples. -// XXX: optimize by avoiding ring buffer usage -void ff_mpa_synth_filter(int16 *synth_buf_ptr, int *synth_buf_offset, - int16 *window, int *dither_state, - int16 *samples, int incr, - int32 sb_samples[32]) -{ - int16 *synth_buf; - const int16 *w, *w2, *p; - int j, offset; - int16 *samples2; - int32 tmp[32]; - int sum, sum2; - int tmp_s; - - offset = *synth_buf_offset; - synth_buf = synth_buf_ptr + offset; - - dct32(tmp, sb_samples); - for(j = 0; j < 32; j++) { - // NOTE: can cause a loss in precision if very high amplitude sound - if (tmp[j] < (-0x7fff - 1)) - synth_buf[j] = (-0x7fff - 1); - else if (tmp[j] > 0x7fff) - synth_buf[j] = 0x7fff; - else - synth_buf[j] = tmp[j]; - } - - // copy to avoid wrap - memcpy(synth_buf + 512, synth_buf, 32 * sizeof(int16)); - - samples2 = samples + 31 * incr; - w = window; - w2 = window + 31; - - sum = *dither_state; - p = synth_buf + 16; - SUM8(MACS, sum, w, p); - p = synth_buf + 48; - SUM8(MLSS, sum, w + 32, p); - *samples = round_sample(&sum); - samples += incr; - w++; - - // we calculate two samples at the same time to avoid one memory - // access per two sample - for(j = 1; j < 16; j++) { - sum2 = 0; - p = synth_buf + 16 + j; - SUM8P2(sum, MACS, sum2, MLSS, w, w2, p); - p = synth_buf + 48 - j; - SUM8P2(sum, MLSS, sum2, MLSS, w + 32, w2 + 32, p); - - *samples = round_sample(&sum); - samples += incr; - sum += sum2; - *samples2 = round_sample(&sum); - samples2 -= incr; - w++; - w2--; - } - - p = synth_buf + 32; - SUM8(MLSS, sum, w + 32, p); - *samples = round_sample(&sum); - *dither_state= sum; - - offset = (offset - 32) & 511; - *synth_buf_offset = offset; -} - -/** - * parses a vlc code, faster then get_vlc() - * @param bits is the number of bits which will be read at once, must be - * identical to nb_bits in init_vlc() - * @param max_depth is the number of times bits bits must be read to completely - * read the longest vlc code - * = (max_vlc_length + bits - 1) / bits - */ -static int getVlc2(GetBitContext *s, int16 (*table)[2], int bits, int maxDepth) { - int reIndex; - int reCache; - int index; - int code; - int n; - - debug(1, "int getVlc2(GetBitContext *s, int16 (*table)[2], int bits, int maxDepth)"); - - reIndex = s->index; - reCache = READ_LE_UINT32(s->buffer + (reIndex >> 3)) >> (reIndex & 0x07); - index = reCache & (0xffffffff >> (32 - bits)); - code = table[index][0]; - n = table[index][1]; - - debug(1, "reIndex : %d", reIndex); - debug(1, "reCache : %d", reCache); - debug(1, "index : %d", index); - debug(1, "code : %d", code); - debug(1, "n : %d", n); - - if (maxDepth > 1 && n < 0){ - reIndex += bits; - reCache = READ_LE_UINT32(s->buffer + (reIndex >> 3)) >> (reIndex & 0x07); - - int nbBits = -n; - - index = (reCache & (0xffffffff >> (32 - nbBits))) + code; - code = table[index][0]; - n = table[index][1]; - - if(maxDepth > 2 && n < 0) { - reIndex += nbBits; - reCache = READ_LE_UINT32(s->buffer + (reIndex >> 3)) >> (reIndex & 0x07); - - nbBits = -n; - - index = (reCache & (0xffffffff >> (32 - nbBits))) + code; - code = table[index][0]; - n = table[index][1]; - } - } - - reCache >>= n; - s->index = reIndex + n; - return code; -} - -static int allocTable(VLC *vlc, int size, int use_static) { - int index; - index = vlc->table_size; - vlc->table_size += size; - if (vlc->table_size > vlc->table_allocated) { - if(use_static) - error("QDM2 cant do anything, init_vlc() is used with too little memory"); - vlc->table_allocated += (1 << vlc->bits); - vlc->table = (int16 (*)[2])realloc(vlc->table, sizeof(int16 *) * 2 * vlc->table_allocated); - if (!vlc->table) - return -1; - } - return index; -} - -#define GET_DATA(v, table, i, wrap, size)\ -{\ - const uint8 *ptr = (const uint8 *)table + i * wrap;\ - switch(size) {\ - case 1:\ - v = *(const uint8 *)ptr;\ - break;\ - case 2:\ - v = *(const uint16 *)ptr;\ - break;\ - default:\ - v = *(const uint32 *)ptr;\ - break;\ - }\ -} - -static int build_table(VLC *vlc, int table_nb_bits, - int nb_codes, - const void *bits, int bits_wrap, int bits_size, - const void *codes, int codes_wrap, int codes_size, - const void *symbols, int symbols_wrap, int symbols_size, - int code_prefix, int n_prefix, int flags) -{ - int i, j, k, n, table_size, table_index, nb, n1, index, code_prefix2, symbol; - uint32 code; - int16 (*table)[2]; - - table_size = 1 << table_nb_bits; - table_index = allocTable(vlc, table_size, flags & 4); - debug(2, "QDM2 new table index=%d size=%d code_prefix=%x n=%d", table_index, table_size, code_prefix, n_prefix); - if (table_index < 0) - return -1; - table = &vlc->table[table_index]; - - for(i = 0; i < table_size; i++) { - table[i][1] = 0; //bits - table[i][0] = -1; //codes - } - - // first pass: map codes and compute auxillary table sizes - for(i = 0; i < nb_codes; i++) { - GET_DATA(n, bits, i, bits_wrap, bits_size); - GET_DATA(code, codes, i, codes_wrap, codes_size); - // we accept tables with holes - if (n <= 0) - continue; - if (!symbols) - symbol = i; - else - GET_DATA(symbol, symbols, i, symbols_wrap, symbols_size); - debug(2, "QDM2 i=%d n=%d code=0x%x", i, n, code); - // if code matches the prefix, it is in the table - n -= n_prefix; - if(flags & 2) - code_prefix2= code & (n_prefix>=32 ? 0xffffffff : (1 << n_prefix)-1); - else - code_prefix2= code >> n; - if (n > 0 && code_prefix2 == code_prefix) { - if (n <= table_nb_bits) { - // no need to add another table - j = (code << (table_nb_bits - n)) & (table_size - 1); - nb = 1 << (table_nb_bits - n); - for(k = 0; k < nb; k++) { - if(flags & 2) - j = (code >> n_prefix) + (k<<n); - debug(2, "QDM2 %4x: code=%d n=%d",j, i, n); - if (table[j][1] /*bits*/ != 0) { - error("QDM2 incorrect codes"); - return -1; - } - table[j][1] = n; //bits - table[j][0] = symbol; - j++; - } - } else { - n -= table_nb_bits; - j = (code >> ((flags & 2) ? n_prefix : n)) & ((1 << table_nb_bits) - 1); - debug(2, "QDM2 %4x: n=%d (subtable)", j, n); - // compute table size - n1 = -table[j][1]; //bits - if (n > n1) - n1 = n; - table[j][1] = -n1; //bits - } - } - } - - // second pass : fill auxillary tables recursively - for(i = 0;i < table_size; i++) { - n = table[i][1]; //bits - if (n < 0) { - n = -n; - if (n > table_nb_bits) { - n = table_nb_bits; - table[i][1] = -n; //bits - } - index = build_table(vlc, n, nb_codes, - bits, bits_wrap, bits_size, - codes, codes_wrap, codes_size, - symbols, symbols_wrap, symbols_size, - (flags & 2) ? (code_prefix | (i << n_prefix)) : ((code_prefix << table_nb_bits) | i), - n_prefix + table_nb_bits, flags); - if (index < 0) - return -1; - // note: realloc has been done, so reload tables - table = &vlc->table[table_index]; - table[i][0] = index; //code - } - } - return table_index; -} - -/* Build VLC decoding tables suitable for use with get_vlc(). - - 'nb_bits' set thee decoding table size (2^nb_bits) entries. The - bigger it is, the faster is the decoding. But it should not be too - big to save memory and L1 cache. '9' is a good compromise. - - 'nb_codes' : number of vlcs codes - - 'bits' : table which gives the size (in bits) of each vlc code. - - 'codes' : table which gives the bit pattern of of each vlc code. - - 'symbols' : table which gives the values to be returned from get_vlc(). - - 'xxx_wrap' : give the number of bytes between each entry of the - 'bits' or 'codes' tables. - - 'xxx_size' : gives the number of bytes of each entry of the 'bits' - or 'codes' tables. - - 'wrap' and 'size' allows to use any memory configuration and types - (byte/word/long) to store the 'bits', 'codes', and 'symbols' tables. - - 'use_static' should be set to 1 for tables, which should be freed - with av_free_static(), 0 if free_vlc() will be used. -*/ -void initVlcSparse(VLC *vlc, int nb_bits, int nb_codes, - const void *bits, int bits_wrap, int bits_size, - const void *codes, int codes_wrap, int codes_size, - const void *symbols, int symbols_wrap, int symbols_size) { - vlc->bits = nb_bits; - - if(vlc->table_size && vlc->table_size == vlc->table_allocated) { - return; - } else if(vlc->table_size) { - error("called on a partially initialized table"); - } - - debug(2, "QDM2 build table nb_codes=%d", nb_codes); - - if (build_table(vlc, nb_bits, nb_codes, - bits, bits_wrap, bits_size, - codes, codes_wrap, codes_size, - symbols, symbols_wrap, symbols_size, - 0, 0, 4 | 2) < 0) { - free(&vlc->table); - return; // Error - } - - if(vlc->table_size != vlc->table_allocated) - error("QDM2 needed %d had %d", vlc->table_size, vlc->table_allocated); -} - -void QDM2Stream::softclipTableInit(void) { - uint16 i; - double dfl = SOFTCLIP_THRESHOLD - 32767; - float delta = 1.0 / -dfl; - - for (i = 0; i < ARRAYSIZE(_softclipTable); i++) - _softclipTable[i] = SOFTCLIP_THRESHOLD - ((int)(sin((float)i * delta) * dfl) & 0x0000FFFF); -} - -// random generated table -void QDM2Stream::rndTableInit(void) { - uint16 i; - uint16 j; - uint32 ldw, hdw; - // TODO: Replace Code with uint64 less version... - int64_t tmp64_1; - int64_t random_seed = 0; - float delta = 1.0 / 16384.0; - - for(i = 0; i < ARRAYSIZE(_noiseTable); i++) { - random_seed = random_seed * 214013 + 2531011; - _noiseTable[i] = (delta * (float)(((int32)random_seed >> 16) & 0x00007FFF)- 1.0) * 1.3; - } - - for (i = 0; i < 256; i++) { - random_seed = 81; - ldw = i; - for (j = 0; j < 5; j++) { - _randomDequantIndex[i][j] = (uint8)((ldw / random_seed) & 0xFF); - ldw = (uint32)ldw % (uint32)random_seed; - tmp64_1 = (random_seed * 0x55555556); - hdw = (uint32)(tmp64_1 >> 32); - random_seed = (int64_t)(hdw + (ldw >> 31)); - } - } - - for (i = 0; i < 128; i++) { - random_seed = 25; - ldw = i; - for (j = 0; j < 3; j++) { - _randomDequantType24[i][j] = (uint8)((ldw / random_seed) & 0xFF); - ldw = (uint32)ldw % (uint32)random_seed; - tmp64_1 = (random_seed * 0x66666667); - hdw = (uint32)(tmp64_1 >> 33); - random_seed = hdw + (ldw >> 31); - } - } -} - -void QDM2Stream::initNoiseSamples(void) { - uint16 i; - uint32 random_seed = 0; - float delta = 1.0 / 16384.0; - - for (i = 0; i < ARRAYSIZE(_noiseSamples); i++) { - random_seed = random_seed * 214013 + 2531011; - _noiseSamples[i] = (delta * (float)((random_seed >> 16) & 0x00007fff) - 1.0); - } -} - -static const uint16 qdm2_vlc_offs[18] = { - 0, 260, 566, 598, 894, 1166, 1230, 1294, 1678, 1950, 2214, 2278, 2310, 2570, 2834, 3124, 3448, 3838 -}; - -void QDM2Stream::initVlc(void) { - static int16 qdm2_table[3838][2]; - - if (!_vlcsInitialized) { - _vlcTabLevel.table = &qdm2_table[qdm2_vlc_offs[0]]; - _vlcTabLevel.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0]; - _vlcTabLevel.table_size = 0; - initVlcSparse(&_vlcTabLevel, 8, 24, - vlc_tab_level_huffbits, 1, 1, - vlc_tab_level_huffcodes, 2, 2, NULL, 0, 0); - - _vlcTabDiff.table = &qdm2_table[qdm2_vlc_offs[1]]; - _vlcTabDiff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1]; - _vlcTabDiff.table_size = 0; - initVlcSparse(&_vlcTabDiff, 8, 37, - vlc_tab_diff_huffbits, 1, 1, - vlc_tab_diff_huffcodes, 2, 2, NULL, 0, 0); - - _vlcTabRun.table = &qdm2_table[qdm2_vlc_offs[2]]; - _vlcTabRun.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2]; - _vlcTabRun.table_size = 0; - initVlcSparse(&_vlcTabRun, 5, 6, - vlc_tab_run_huffbits, 1, 1, - vlc_tab_run_huffcodes, 1, 1, NULL, 0, 0); - - _fftLevelExpAltVlc.table = &qdm2_table[qdm2_vlc_offs[3]]; - _fftLevelExpAltVlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3]; - _fftLevelExpAltVlc.table_size = 0; - initVlcSparse(&_fftLevelExpAltVlc, 8, 28, - fft_level_exp_alt_huffbits, 1, 1, - fft_level_exp_alt_huffcodes, 2, 2, NULL, 0, 0); - - _fftLevelExpVlc.table = &qdm2_table[qdm2_vlc_offs[4]]; - _fftLevelExpVlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4]; - _fftLevelExpVlc.table_size = 0; - initVlcSparse(&_fftLevelExpVlc, 8, 20, - fft_level_exp_huffbits, 1, 1, - fft_level_exp_huffcodes, 2, 2, NULL, 0, 0); - - _fftStereoExpVlc.table = &qdm2_table[qdm2_vlc_offs[5]]; - _fftStereoExpVlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5]; - _fftStereoExpVlc.table_size = 0; - initVlcSparse(&_fftStereoExpVlc, 6, 7, - fft_stereo_exp_huffbits, 1, 1, - fft_stereo_exp_huffcodes, 1, 1, NULL, 0, 0); - - _fftStereoPhaseVlc.table = &qdm2_table[qdm2_vlc_offs[6]]; - _fftStereoPhaseVlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6]; - _fftStereoPhaseVlc.table_size = 0; - initVlcSparse(&_fftStereoPhaseVlc, 6, 9, - fft_stereo_phase_huffbits, 1, 1, - fft_stereo_phase_huffcodes, 1, 1, NULL, 0, 0); - - _vlcTabToneLevelIdxHi1.table = &qdm2_table[qdm2_vlc_offs[7]]; - _vlcTabToneLevelIdxHi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7]; - _vlcTabToneLevelIdxHi1.table_size = 0; - initVlcSparse(&_vlcTabToneLevelIdxHi1, 8, 20, - vlc_tab_tone_level_idx_hi1_huffbits, 1, 1, - vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, NULL, 0, 0); - - _vlcTabToneLevelIdxMid.table = &qdm2_table[qdm2_vlc_offs[8]]; - _vlcTabToneLevelIdxMid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8]; - _vlcTabToneLevelIdxMid.table_size = 0; - initVlcSparse(&_vlcTabToneLevelIdxMid, 8, 24, - vlc_tab_tone_level_idx_mid_huffbits, 1, 1, - vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, NULL, 0, 0); - - _vlcTabToneLevelIdxHi2.table = &qdm2_table[qdm2_vlc_offs[9]]; - _vlcTabToneLevelIdxHi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9]; - _vlcTabToneLevelIdxHi2.table_size = 0; - initVlcSparse(&_vlcTabToneLevelIdxHi2, 8, 24, - vlc_tab_tone_level_idx_hi2_huffbits, 1, 1, - vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, NULL, 0, 0); - - _vlcTabType30.table = &qdm2_table[qdm2_vlc_offs[10]]; - _vlcTabType30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10]; - _vlcTabType30.table_size = 0; - initVlcSparse(&_vlcTabType30, 6, 9, - vlc_tab_type30_huffbits, 1, 1, - vlc_tab_type30_huffcodes, 1, 1, NULL, 0, 0); - - _vlcTabType34.table = &qdm2_table[qdm2_vlc_offs[11]]; - _vlcTabType34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11]; - _vlcTabType34.table_size = 0; - initVlcSparse(&_vlcTabType34, 5, 10, - vlc_tab_type34_huffbits, 1, 1, - vlc_tab_type34_huffcodes, 1, 1, NULL, 0, 0); - - _vlcTabFftToneOffset[0].table = &qdm2_table[qdm2_vlc_offs[12]]; - _vlcTabFftToneOffset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12]; - _vlcTabFftToneOffset[0].table_size = 0; - initVlcSparse(&_vlcTabFftToneOffset[0], 8, 23, - vlc_tab_fft_tone_offset_0_huffbits, 1, 1, - vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, NULL, 0, 0); - - _vlcTabFftToneOffset[1].table = &qdm2_table[qdm2_vlc_offs[13]]; - _vlcTabFftToneOffset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13]; - _vlcTabFftToneOffset[1].table_size = 0; - initVlcSparse(&_vlcTabFftToneOffset[1], 8, 28, - vlc_tab_fft_tone_offset_1_huffbits, 1, 1, - vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, NULL, 0, 0); - - _vlcTabFftToneOffset[2].table = &qdm2_table[qdm2_vlc_offs[14]]; - _vlcTabFftToneOffset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14]; - _vlcTabFftToneOffset[2].table_size = 0; - initVlcSparse(&_vlcTabFftToneOffset[2], 8, 32, - vlc_tab_fft_tone_offset_2_huffbits, 1, 1, - vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, NULL, 0, 0); - - _vlcTabFftToneOffset[3].table = &qdm2_table[qdm2_vlc_offs[15]]; - _vlcTabFftToneOffset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15]; - _vlcTabFftToneOffset[3].table_size = 0; - initVlcSparse(&_vlcTabFftToneOffset[3], 8, 35, - vlc_tab_fft_tone_offset_3_huffbits, 1, 1, - vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, NULL, 0, 0); - - _vlcTabFftToneOffset[4].table = &qdm2_table[qdm2_vlc_offs[16]]; - _vlcTabFftToneOffset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16]; - _vlcTabFftToneOffset[4].table_size = 0; - initVlcSparse(&_vlcTabFftToneOffset[4], 8, 38, - vlc_tab_fft_tone_offset_4_huffbits, 1, 1, - vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, NULL, 0, 0); - - _vlcsInitialized = true; - } -} - -QDM2Stream::QDM2Stream(Common::SeekableReadStream *stream, Common::SeekableReadStream *extraData) { - uint32 tmp; - int32 tmp_s; - int tmp_val; - int i; - - debug(1, "QDM2Stream::QDM2Stream() Call"); - - _stream = stream; - _compressedData = NULL; - _subPacket = 0; - memset(_quantizedCoeffs, 0, sizeof(_quantizedCoeffs)); - memset(_fftLevelExp, 0, sizeof(_fftLevelExp)); - _noiseIdx = 0; - memset(_fftCoefsMinIndex, 0, sizeof(_fftCoefsMinIndex)); - memset(_fftCoefsMaxIndex, 0, sizeof(_fftCoefsMaxIndex)); - _fftToneStart = 0; - _fftToneEnd = 0; - for(i = 0; i < ARRAYSIZE(_subPacketListA); i++) { - _subPacketListA[i].packet = NULL; - _subPacketListA[i].next = NULL; - } - _subPacketsB = 0; - for(i = 0; i < ARRAYSIZE(_subPacketListB); i++) { - _subPacketListB[i].packet = NULL; - _subPacketListB[i].next = NULL; - } - for(i = 0; i < ARRAYSIZE(_subPacketListC); i++) { - _subPacketListC[i].packet = NULL; - _subPacketListC[i].next = NULL; - } - for(i = 0; i < ARRAYSIZE(_subPacketListD); i++) { - _subPacketListD[i].packet = NULL; - _subPacketListD[i].next = NULL; - } - memset(_synthBuf, 0, sizeof(_synthBuf)); - memset(_synthBufOffset, 0, sizeof(_synthBufOffset)); - memset(_sbSamples, 0, sizeof(_sbSamples)); - memset(_outputBuffer, 0, sizeof(_outputBuffer)); - _vlcsInitialized = false; - _superblocktype_2_3 = 0; - _hasErrors = false; - - // Rewind extraData stream from any previous calls... - extraData->seek(0, SEEK_SET); - - tmp_s = extraData->readSint32BE(); - debug(1, "QDM2Stream::QDM2Stream() extraSize: %d", tmp_s); - if ((extraData->size() - extraData->pos()) / 4 + 1 != tmp_s) - warning("QDM2Stream::QDM2Stream() extraSize mismatch - Expected %d", (extraData->size() - extraData->pos()) / 4 + 1); - if (tmp_s < 12) - error("QDM2Stream::QDM2Stream() Insufficient extraData"); - - tmp = extraData->readUint32BE(); - debug(1, "QDM2Stream::QDM2Stream() extraTag: %d", tmp); - if (tmp != MKID_BE('frma')) - warning("QDM2Stream::QDM2Stream() extraTag mismatch"); - - tmp = extraData->readUint32BE(); - debug(1, "QDM2Stream::QDM2Stream() extraType: %d", tmp); - if (tmp == MKID_BE('QDMC')) - warning("QDM2Stream::QDM2Stream() QDMC stream type not supported."); - else if (tmp != MKID_BE('QDM2')) - error("QDM2Stream::QDM2Stream() Unsupported stream type"); - - tmp_s = extraData->readSint32BE(); - debug(1, "QDM2Stream::QDM2Stream() extraSize2: %d", tmp_s); - if ((extraData->size() - extraData->pos()) + 4 != tmp_s) - warning("QDM2Stream::QDM2Stream() extraSize2 mismatch - Expected %d", (extraData->size() - extraData->pos()) + 4); - - tmp = extraData->readUint32BE(); - debug(1, "QDM2Stream::QDM2Stream() extraTag2: %d", tmp); - if (tmp != MKID_BE('QDCA')) - warning("QDM2Stream::QDM2Stream() extraTag2 mismatch"); - - if (extraData->readUint32BE() != 1) - warning("QDM2Stream::QDM2Stream() u0 field not 1"); - - _channels = extraData->readUint32BE(); - debug(1, "QDM2Stream::QDM2Stream() channels: %d", _channels); - - _sampleRate = extraData->readUint32BE(); - debug(1, "QDM2Stream::QDM2Stream() sampleRate: %d", _sampleRate); - - _bitRate = extraData->readUint32BE(); - debug(1, "QDM2Stream::QDM2Stream() bitRate: %d", _bitRate); - - _blockSize = extraData->readUint32BE(); - debug(1, "QDM2Stream::QDM2Stream() blockSize: %d", _blockSize); - - _frameSize = extraData->readUint32BE(); - debug(1, "QDM2Stream::QDM2Stream() frameSize: %d", _frameSize); - - _packetSize = extraData->readUint32BE(); - debug(1, "QDM2Stream::QDM2Stream() packetSize: %d", _packetSize); - - if (extraData->size() - extraData->pos() != 0) { - tmp_s = extraData->readSint32BE(); - debug(1, "QDM2Stream::QDM2Stream() extraSize3: %d", tmp_s); - if (extraData->size() + 4 != tmp_s) - warning("QDM2Stream::QDM2Stream() extraSize3 mismatch - Expected %d", extraData->size() + 4); - - tmp = extraData->readUint32BE(); - debug(1, "QDM2Stream::QDM2Stream() extraTag3: %d", tmp); - if (tmp != MKID_BE('QDCP')) - warning("QDM2Stream::QDM2Stream() extraTag3 mismatch"); - - if ((float)extraData->readUint32BE() != 1.0) - warning("QDM2Stream::QDM2Stream() uf0 field not 1.0"); - - if (extraData->readUint32BE() != 0) - warning("QDM2Stream::QDM2Stream() u1 field not 0"); - - if ((float)extraData->readUint32BE() != 1.0) - warning("QDM2Stream::QDM2Stream() uf1 field not 1.0"); - - if ((float)extraData->readUint32BE() != 1.0) - warning("QDM2Stream::QDM2Stream() uf2 field not 1.0"); - - if (extraData->readUint32BE() != 27) - warning("QDM2Stream::QDM2Stream() u2 field not 27"); - - if (extraData->readUint32BE() != 8) - warning("QDM2Stream::QDM2Stream() u3 field not 8"); - - if (extraData->readUint32BE() != 0) - warning("QDM2Stream::QDM2Stream() u4 field not 0"); - } - - _fftOrder = scummvm_log2(_frameSize) + 1; - _fftFrameSize = 2 * _frameSize; // complex has two floats - - // something like max decodable tones - _groupOrder = scummvm_log2(_blockSize) + 1; - _sFrameSize = _blockSize / 16; // 16 iterations per super block - - _subSampling = _fftOrder - 7; - _frequencyRange = 255 / (1 << (2 - _subSampling)); - - switch ((_subSampling * 2 + _channels - 1)) { - case 0: - tmp = 40; - break; - case 1: - tmp = 48; - break; - case 2: - tmp = 56; - break; - case 3: - tmp = 72; - break; - case 4: - tmp = 80; - break; - case 5: - tmp = 100; - break; - default: - tmp = _subSampling; - break; - } - - tmp_val = 0; - if ((tmp * 1000) < _bitRate) tmp_val = 1; - if ((tmp * 1440) < _bitRate) tmp_val = 2; - if ((tmp * 1760) < _bitRate) tmp_val = 3; - if ((tmp * 2240) < _bitRate) tmp_val = 4; - _cmTableSelect = tmp_val; - - if (_subSampling == 0) - tmp = 7999; - else - tmp = ((-(_subSampling -1)) & 8000) + 20000; - - if (tmp < 8000) - _coeffPerSbSelect = 0; - else if (tmp <= 16000) - _coeffPerSbSelect = 1; - else - _coeffPerSbSelect = 2; - - if (_fftOrder < 7 || _fftOrder > 9) - error("QDM2Stream::QDM2Stream() Unsupported fft_order: %d", _fftOrder); - - rdftInit(&_rdftCtx, _fftOrder, IRDFT); - - initVlc(); - ff_mpa_synth_init(ff_mpa_synth_window); - softclipTableInit(); - rndTableInit(); - initNoiseSamples(); - - _compressedData = new uint8[_packetSize]; -} - -QDM2Stream::~QDM2Stream() { - delete[] _compressedData; - delete _stream; -} - -static int qdm2_get_vlc(GetBitContext *gb, VLC *vlc, int flag, int depth) { - int value = getVlc2(gb, vlc->table, vlc->bits, depth); - - // stage-2, 3 bits exponent escape sequence - if (value-- == 0) - value = getBits(gb, getBits (gb, 3) + 1); - - // stage-3, optional - if (flag) { - int tmp = vlc_stage3_values[value]; - - if ((value & ~3) > 0) - tmp += getBits(gb, (value >> 2)); - value = tmp; - } - - return value; -} - -static int qdm2_get_se_vlc(VLC *vlc, GetBitContext *gb, int depth) -{ - int value = qdm2_get_vlc(gb, vlc, 0, depth); - - return (value & 1) ? ((value + 1) >> 1) : -(value >> 1); -} - -/** - * QDM2 checksum - * - * @param data pointer to data to be checksum'ed - * @param length data length - * @param value checksum value - * - * @return 0 if checksum is OK - */ -static uint16 qdm2_packet_checksum(const uint8 *data, int length, int value) { - int i; - - for (i = 0; i < length; i++) - value -= data[i]; - - return (uint16)(value & 0xffff); -} - -/** - * Fills a QDM2SubPacket structure with packet type, size, and data pointer. - * - * @param gb bitreader context - * @param sub_packet packet under analysis - */ -static void qdm2_decode_sub_packet_header(GetBitContext *gb, QDM2SubPacket *sub_packet) -{ - sub_packet->type = getBits (gb, 8); - - if (sub_packet->type == 0) { - sub_packet->size = 0; - sub_packet->data = NULL; - } else { - sub_packet->size = getBits (gb, 8); - - if (sub_packet->type & 0x80) { - sub_packet->size <<= 8; - sub_packet->size |= getBits (gb, 8); - sub_packet->type &= 0x7f; - } - - if (sub_packet->type == 0x7f) - sub_packet->type |= (getBits (gb, 8) << 8); - - sub_packet->data = &gb->buffer[getBitsCount(gb) / 8]; // FIXME: this depends on bitreader internal data - } - - debug(1, "QDM2 Subpacket: type=%d size=%d start_offs=%x", sub_packet->type, sub_packet->size, getBitsCount(gb) / 8); -} - -/** - * Return node pointer to first packet of requested type in list. - * - * @param list list of subpackets to be scanned - * @param type type of searched subpacket - * @return node pointer for subpacket if found, else NULL - */ -static QDM2SubPNode* qdm2_search_subpacket_type_in_list(QDM2SubPNode *list, int type) -{ - while (list != NULL && list->packet != NULL) { - if (list->packet->type == type) - return list; - list = list->next; - } - return NULL; -} - -/** - * Replaces 8 elements with their average value. - * Called by qdm2_decode_superblock before starting subblock decoding. - */ -void QDM2Stream::average_quantized_coeffs(void) { - int i, j, n, ch, sum; - - n = coeff_per_sb_for_avg[_coeffPerSbSelect][QDM2_SB_USED(_subSampling) - 1] + 1; - - for (ch = 0; ch < _channels; ch++) { - for (i = 0; i < n; i++) { - sum = 0; - - for (j = 0; j < 8; j++) - sum += _quantizedCoeffs[ch][i][j]; - - sum /= 8; - if (sum > 0) - sum--; - - for (j = 0; j < 8; j++) - _quantizedCoeffs[ch][i][j] = sum; - } - } -} - -/** - * Build subband samples with noise weighted by q->tone_level. - * Called by synthfilt_build_sb_samples. - * - * @param sb subband index - */ -void QDM2Stream::build_sb_samples_from_noise(int sb) { - int ch, j; - - FIX_NOISE_IDX(_noiseIdx); - - if (!_channels) - return; - - for (ch = 0; ch < _channels; ch++) { - for (j = 0; j < 64; j++) { - _sbSamples[ch][j * 2][sb] = (int32)(SB_DITHERING_NOISE(sb, _noiseIdx) * _toneLevel[ch][sb][j] + .5); - _sbSamples[ch][j * 2 + 1][sb] = (int32)(SB_DITHERING_NOISE(sb, _noiseIdx) * _toneLevel[ch][sb][j] + .5); - } - } -} - -/** - * Called while processing data from subpackets 11 and 12. - * Used after making changes to coding_method array. - * - * @param sb subband index - * @param channels number of channels - * @param coding_method q->coding_method[0][0][0] - */ -void QDM2Stream::fix_coding_method_array(int sb, int channels, sb_int8_array coding_method) -{ - int j, k; - int ch; - int run, case_val; - int switchtable[23] = {0,5,1,5,5,5,5,5,2,5,5,5,5,5,5,5,3,5,5,5,5,5,4}; - - for (ch = 0; ch < channels; ch++) { - for (j = 0; j < 64; ) { - if((coding_method[ch][sb][j] - 8) > 22) { - run = 1; - case_val = 8; - } else { - switch (switchtable[coding_method[ch][sb][j]-8]) { - case 0: run = 10; case_val = 10; break; - case 1: run = 1; case_val = 16; break; - case 2: run = 5; case_val = 24; break; - case 3: run = 3; case_val = 30; break; - case 4: run = 1; case_val = 30; break; - case 5: run = 1; case_val = 8; break; - default: run = 1; case_val = 8; break; - } - } - for (k = 0; k < run; k++) - if (j + k < 128) - if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j]) - if (k > 0) { - warning("QDM2 Untested Code: not debugged, almost never used"); - memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8)); - memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8)); - } - j += run; - } - } -} - -/** - * Related to synthesis filter - * Called by process_subpacket_10 - * - * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10 - */ -void QDM2Stream::fill_tone_level_array(int flag) { - int i, sb, ch, sb_used; - int tmp, tab; - - // This should never happen - if (_channels <= 0) - return; - - for (ch = 0; ch < _channels; ch++) { - for (sb = 0; sb < 30; sb++) { - for (i = 0; i < 8; i++) { - if ((tab=coeff_per_sb_for_dequant[_coeffPerSbSelect][sb]) < (last_coeff[_coeffPerSbSelect] - 1)) - tmp = _quantizedCoeffs[ch][tab + 1][i] * dequant_table[_coeffPerSbSelect][tab + 1][sb]+ - _quantizedCoeffs[ch][tab][i] * dequant_table[_coeffPerSbSelect][tab][sb]; - else - tmp = _quantizedCoeffs[ch][tab][i] * dequant_table[_coeffPerSbSelect][tab][sb]; - if(tmp < 0) - tmp += 0xff; - _toneLevelIdxBase[ch][sb][i] = (tmp / 256) & 0xff; - } - } - } - - sb_used = QDM2_SB_USED(_subSampling); - - if ((_superblocktype_2_3 != 0) && !flag) { - for (sb = 0; sb < sb_used; sb++) { - for (ch = 0; ch < _channels; ch++) { - for (i = 0; i < 64; i++) { - _toneLevelIdx[ch][sb][i] = _toneLevelIdxBase[ch][sb][i / 8]; - if (_toneLevelIdx[ch][sb][i] < 0) - _toneLevel[ch][sb][i] = 0; - else - _toneLevel[ch][sb][i] = fft_tone_level_table[0][_toneLevelIdx[ch][sb][i] & 0x3f]; - } - } - } - } else { - tab = _superblocktype_2_3 ? 0 : 1; - for (sb = 0; sb < sb_used; sb++) { - if ((sb >= 4) && (sb <= 23)) { - for (ch = 0; ch < _channels; ch++) { - for (i = 0; i < 64; i++) { - tmp = _toneLevelIdxBase[ch][sb][i / 8] - - _toneLevelIdxHi1[ch][sb / 8][i / 8][i % 8] - - _toneLevelIdxMid[ch][sb - 4][i / 8] - - _toneLevelIdxHi2[ch][sb - 4]; - _toneLevelIdx[ch][sb][i] = tmp & 0xff; - if ((tmp < 0) || (!_superblocktype_2_3 && !tmp)) - _toneLevel[ch][sb][i] = 0; - else - _toneLevel[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; - } - } - } else { - if (sb > 4) { - for (ch = 0; ch < _channels; ch++) { - for (i = 0; i < 64; i++) { - tmp = _toneLevelIdxBase[ch][sb][i / 8] - - _toneLevelIdxHi1[ch][2][i / 8][i % 8] - - _toneLevelIdxHi2[ch][sb - 4]; - _toneLevelIdx[ch][sb][i] = tmp & 0xff; - if ((tmp < 0) || (!_superblocktype_2_3 && !tmp)) - _toneLevel[ch][sb][i] = 0; - else - _toneLevel[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; - } - } - } else { - for (ch = 0; ch < _channels; ch++) { - for (i = 0; i < 64; i++) { - tmp = _toneLevelIdx[ch][sb][i] = _toneLevelIdxBase[ch][sb][i / 8]; - if ((tmp < 0) || (!_superblocktype_2_3 && !tmp)) - _toneLevel[ch][sb][i] = 0; - else - _toneLevel[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; - } - } - } - } - } - } -} - -/** - * Related to synthesis filter - * Called by process_subpacket_11 - * c is built with data from subpacket 11 - * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples - * - * @param tone_level_idx - * @param tone_level_idx_temp - * @param coding_method q->coding_method[0][0][0] - * @param nb_channels number of channels - * @param c coming from subpacket 11, passed as 8*c - * @param superblocktype_2_3 flag based on superblock packet type - * @param cm_table_select q->cm_table_select - */ -void QDM2Stream::fill_coding_method_array(sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp, - sb_int8_array coding_method, int nb_channels, - int c, int superblocktype_2_3, int cm_table_select) { - int ch, sb, j; - int tmp, acc, esp_40, comp; - int add1, add2, add3, add4; - // TODO : Remove multres 64 bit variable necessity... - int64_t multres; - - // This should never happen - if (nb_channels <= 0) - return; - if (!superblocktype_2_3) { - warning("QDM2 This case is untested, no samples available"); - for (ch = 0; ch < nb_channels; ch++) - for (sb = 0; sb < 30; sb++) { - for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer - add1 = tone_level_idx[ch][sb][j] - 10; - if (add1 < 0) - add1 = 0; - add2 = add3 = add4 = 0; - if (sb > 1) { - add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6; - if (add2 < 0) - add2 = 0; - } - if (sb > 0) { - add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6; - if (add3 < 0) - add3 = 0; - } - if (sb < 29) { - add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6; - if (add4 < 0) - add4 = 0; - } - tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1; - if (tmp < 0) - tmp = 0; - tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff; - } - tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1]; - } - acc = 0; - for (ch = 0; ch < nb_channels; ch++) - for (sb = 0; sb < 30; sb++) - for (j = 0; j < 64; j++) - acc += tone_level_idx_temp[ch][sb][j]; - - multres = 0x66666667 * (acc * 10); - esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31); - for (ch = 0; ch < nb_channels; ch++) - for (sb = 0; sb < 30; sb++) - for (j = 0; j < 64; j++) { - comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10; - if (comp < 0) - comp += 0xff; - comp /= 256; // signed shift - switch(sb) { - case 0: - if (comp < 30) - comp = 30; - comp += 15; - break; - case 1: - if (comp < 24) - comp = 24; - comp += 10; - break; - case 2: - case 3: - case 4: - if (comp < 16) - comp = 16; - } - if (comp <= 5) - tmp = 0; - else if (comp <= 10) - tmp = 10; - else if (comp <= 16) - tmp = 16; - else if (comp <= 24) - tmp = -1; - else - tmp = 0; - coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff; - } - for (sb = 0; sb < 30; sb++) - fix_coding_method_array(sb, nb_channels, coding_method); - for (ch = 0; ch < nb_channels; ch++) - for (sb = 0; sb < 30; sb++) - for (j = 0; j < 64; j++) - if (sb >= 10) { - if (coding_method[ch][sb][j] < 10) - coding_method[ch][sb][j] = 10; - } else { - if (sb >= 2) { - if (coding_method[ch][sb][j] < 16) - coding_method[ch][sb][j] = 16; - } else { - if (coding_method[ch][sb][j] < 30) - coding_method[ch][sb][j] = 30; - } - } - } else { // superblocktype_2_3 != 0 - for (ch = 0; ch < nb_channels; ch++) - for (sb = 0; sb < 30; sb++) - for (j = 0; j < 64; j++) - coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb]; - } -} - -/** - * - * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8 - * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used - * - * @param gb bitreader context - * @param length packet length in bits - * @param sb_min lower subband processed (sb_min included) - * @param sb_max higher subband processed (sb_max excluded) - */ -void QDM2Stream::synthfilt_build_sb_samples(GetBitContext *gb, int length, int sb_min, int sb_max) { - int sb, j, k, n, ch, run, channels; - int joined_stereo, zero_encoding, chs; - int type34_first; - float type34_div = 0; - float type34_predictor; - float samples[10], sign_bits[16]; - - if (length == 0) { - // If no data use noise - for (sb = sb_min; sb < sb_max; sb++) - build_sb_samples_from_noise(sb); - - return; - } - - for (sb = sb_min; sb < sb_max; sb++) { - FIX_NOISE_IDX(_noiseIdx); - - channels = _channels; - - if (_channels <= 1 || sb < 12) - joined_stereo = 0; - else if (sb >= 24) - joined_stereo = 1; - else - joined_stereo = (BITS_LEFT(length,gb) >= 1) ? getBits1 (gb) : 0; - - if (joined_stereo) { - if (BITS_LEFT(length,gb) >= 16) - for (j = 0; j < 16; j++) - sign_bits[j] = getBits1(gb); - - for (j = 0; j < 64; j++) - if (_codingMethod[1][sb][j] > _codingMethod[0][sb][j]) - _codingMethod[0][sb][j] = _codingMethod[1][sb][j]; - - fix_coding_method_array(sb, _channels, _codingMethod); - channels = 1; - } - - for (ch = 0; ch < channels; ch++) { - zero_encoding = (BITS_LEFT(length,gb) >= 1) ? getBits1(gb) : 0; - type34_predictor = 0.0; - type34_first = 1; - - for (j = 0; j < 128; ) { - switch (_codingMethod[ch][sb][j / 2]) { - case 8: - if (BITS_LEFT(length,gb) >= 10) { - if (zero_encoding) { - for (k = 0; k < 5; k++) { - if ((j + 2 * k) >= 128) - break; - samples[2 * k] = getBits1(gb) ? dequant_1bit[joined_stereo][2 * getBits1(gb)] : 0; - } - } else { - n = getBits(gb, 8); - for (k = 0; k < 5; k++) - samples[2 * k] = dequant_1bit[joined_stereo][_randomDequantIndex[n][k]]; - } - for (k = 0; k < 5; k++) - samples[2 * k + 1] = SB_DITHERING_NOISE(sb, _noiseIdx); - } else { - for (k = 0; k < 10; k++) - samples[k] = SB_DITHERING_NOISE(sb, _noiseIdx); - } - run = 10; - break; - - case 10: - if (BITS_LEFT(length,gb) >= 1) { - double f = 0.81; - - if (getBits1(gb)) - f = -f; - f -= _noiseSamples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0; - samples[0] = f; - } else { - samples[0] = SB_DITHERING_NOISE(sb, _noiseIdx); - } - run = 1; - break; - - case 16: - if (BITS_LEFT(length,gb) >= 10) { - if (zero_encoding) { - for (k = 0; k < 5; k++) { - if ((j + k) >= 128) - break; - samples[k] = (getBits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * getBits1(gb)]; - } - } else { - n = getBits (gb, 8); - for (k = 0; k < 5; k++) - samples[k] = dequant_1bit[joined_stereo][_randomDequantIndex[n][k]]; - } - } else { - for (k = 0; k < 5; k++) - samples[k] = SB_DITHERING_NOISE(sb, _noiseIdx); - } - run = 5; - break; - - case 24: - if (BITS_LEFT(length,gb) >= 7) { - n = getBits(gb, 7); - for (k = 0; k < 3; k++) - samples[k] = (_randomDequantType24[n][k] - 2.0) * 0.5; - } else { - for (k = 0; k < 3; k++) - samples[k] = SB_DITHERING_NOISE(sb, _noiseIdx); - } - run = 3; - break; - - case 30: - if (BITS_LEFT(length,gb) >= 4) - samples[0] = type30_dequant[qdm2_get_vlc(gb, &_vlcTabType30, 0, 1)]; - else - samples[0] = SB_DITHERING_NOISE(sb, _noiseIdx); - - run = 1; - break; - - case 34: - if (BITS_LEFT(length,gb) >= 7) { - if (type34_first) { - type34_div = (float)(1 << getBits(gb, 2)); - samples[0] = ((float)getBits(gb, 5) - 16.0) / 15.0; - type34_predictor = samples[0]; - type34_first = 0; - } else { - samples[0] = type34_delta[qdm2_get_vlc(gb, &_vlcTabType34, 0, 1)] / type34_div + type34_predictor; - type34_predictor = samples[0]; - } - } else { - samples[0] = SB_DITHERING_NOISE(sb, _noiseIdx); - } - run = 1; - break; - - default: - samples[0] = SB_DITHERING_NOISE(sb, _noiseIdx); - run = 1; - break; - } - - if (joined_stereo) { - float tmp[10][MPA_MAX_CHANNELS]; - - for (k = 0; k < run; k++) { - tmp[k][0] = samples[k]; - tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k]; - } - for (chs = 0; chs < _channels; chs++) - for (k = 0; k < run; k++) - if ((j + k) < 128) - _sbSamples[chs][j + k][sb] = (int32)(_toneLevel[chs][sb][((j + k)/2)] * tmp[k][chs] + .5); - } else { - for (k = 0; k < run; k++) - if ((j + k) < 128) - _sbSamples[ch][j + k][sb] = (int32)(_toneLevel[ch][sb][(j + k)/2] * samples[k] + .5); - } - - j += run; - } // j loop - } // channel loop - } // subband loop -} - -/** - * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]). - * This is similar to process_subpacket_9, but for a single channel and for element [0] - * same VLC tables as process_subpacket_9 are used. - * - * @param quantized_coeffs pointer to quantized_coeffs[ch][0] - * @param gb bitreader context - * @param length packet length in bits - */ -void QDM2Stream::init_quantized_coeffs_elem0(int8 *quantized_coeffs, GetBitContext *gb, int length) { - int i, k, run, level, diff; - - if (BITS_LEFT(length,gb) < 16) - return; - level = qdm2_get_vlc(gb, &_vlcTabLevel, 0, 2); - - quantized_coeffs[0] = level; - - for (i = 0; i < 7; ) { - if (BITS_LEFT(length,gb) < 16) - break; - run = qdm2_get_vlc(gb, &_vlcTabRun, 0, 1) + 1; - - if (BITS_LEFT(length,gb) < 16) - break; - diff = qdm2_get_se_vlc(&_vlcTabDiff, gb, 2); - - for (k = 1; k <= run; k++) - quantized_coeffs[i + k] = (level + ((k * diff) / run)); - - level += diff; - i += run; - } -} - -/** - * Related to synthesis filter, process data from packet 10 - * Init part of quantized_coeffs via function init_quantized_coeffs_elem0 - * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10 - * - * @param gb bitreader context - * @param length packet length in bits - */ -void QDM2Stream::init_tone_level_dequantization(GetBitContext *gb, int length) { - int sb, j, k, n, ch; - - for (ch = 0; ch < _channels; ch++) { - init_quantized_coeffs_elem0(_quantizedCoeffs[ch][0], gb, length); - - if (BITS_LEFT(length,gb) < 16) { - memset(_quantizedCoeffs[ch][0], 0, 8); - break; - } - } - - n = _subSampling + 1; - - for (sb = 0; sb < n; sb++) - for (ch = 0; ch < _channels; ch++) - for (j = 0; j < 8; j++) { - if (BITS_LEFT(length,gb) < 1) - break; - if (getBits1(gb)) { - for (k=0; k < 8; k++) { - if (BITS_LEFT(length,gb) < 16) - break; - _toneLevelIdxHi1[ch][sb][j][k] = qdm2_get_vlc(gb, &_vlcTabToneLevelIdxHi1, 0, 2); - } - } else { - for (k=0; k < 8; k++) - _toneLevelIdxHi1[ch][sb][j][k] = 0; - } - } - - n = QDM2_SB_USED(_subSampling) - 4; - - for (sb = 0; sb < n; sb++) - for (ch = 0; ch < _channels; ch++) { - if (BITS_LEFT(length,gb) < 16) - break; - _toneLevelIdxHi2[ch][sb] = qdm2_get_vlc(gb, &_vlcTabToneLevelIdxHi2, 0, 2); - if (sb > 19) - _toneLevelIdxHi2[ch][sb] -= 16; - else - for (j = 0; j < 8; j++) - _toneLevelIdxMid[ch][sb][j] = -16; - } - - n = QDM2_SB_USED(_subSampling) - 5; - - for (sb = 0; sb < n; sb++) { - for (ch = 0; ch < _channels; ch++) { - for (j = 0; j < 8; j++) { - if (BITS_LEFT(length,gb) < 16) - break; - _toneLevelIdxMid[ch][sb][j] = qdm2_get_vlc(gb, &_vlcTabToneLevelIdxMid, 0, 2) - 32; - } - } - } -} - -/** - * Process subpacket 9, init quantized_coeffs with data from it - * - * @param node pointer to node with packet - */ -void QDM2Stream::process_subpacket_9(QDM2SubPNode *node) { - GetBitContext gb; - int i, j, k, n, ch, run, level, diff; - - initGetBits(&gb, node->packet->data, node->packet->size*8); - - n = coeff_per_sb_for_avg[_coeffPerSbSelect][QDM2_SB_USED(_subSampling) - 1] + 1; // same as averagesomething function - - for (i = 1; i < n; i++) - for (ch = 0; ch < _channels; ch++) { - level = qdm2_get_vlc(&gb, &_vlcTabLevel, 0, 2); - _quantizedCoeffs[ch][i][0] = level; - - for (j = 0; j < (8 - 1); ) { - run = qdm2_get_vlc(&gb, &_vlcTabRun, 0, 1) + 1; - diff = qdm2_get_se_vlc(&_vlcTabDiff, &gb, 2); - - for (k = 1; k <= run; k++) - _quantizedCoeffs[ch][i][j + k] = (level + ((k*diff) / run)); - - level += diff; - j += run; - } - } - - for (ch = 0; ch < _channels; ch++) - for (i = 0; i < 8; i++) - _quantizedCoeffs[ch][0][i] = 0; -} - -/** - * Process subpacket 10 if not null, else - * - * @param node pointer to node with packet - * @param length packet length in bits - */ -void QDM2Stream::process_subpacket_10(QDM2SubPNode *node, int length) { - GetBitContext gb; - - initGetBits(&gb, ((node == NULL) ? _emptyBuffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); - - if (length != 0) { - init_tone_level_dequantization(&gb, length); - fill_tone_level_array(1); - } else { - fill_tone_level_array(0); - } -} - -/** - * Process subpacket 11 - * - * @param node pointer to node with packet - * @param length packet length in bit - */ -void QDM2Stream::process_subpacket_11(QDM2SubPNode *node, int length) { - GetBitContext gb; - - initGetBits(&gb, ((node == NULL) ? _emptyBuffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); - if (length >= 32) { - int c = getBits (&gb, 13); - - if (c > 3) - fill_coding_method_array(_toneLevelIdx, _toneLevelIdxTemp, _codingMethod, - _channels, 8*c, _superblocktype_2_3, _cmTableSelect); - } - - synthfilt_build_sb_samples(&gb, length, 0, 8); -} - -/** - * Process subpacket 12 - * - * @param node pointer to node with packet - * @param length packet length in bits - */ -void QDM2Stream::process_subpacket_12(QDM2SubPNode *node, int length) { - GetBitContext gb; - - initGetBits(&gb, ((node == NULL) ? _emptyBuffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); - synthfilt_build_sb_samples(&gb, length, 8, QDM2_SB_USED(_subSampling)); -} - -/* - * Process new subpackets for synthesis filter - * - * @param list list with synthesis filter packets (list D) - */ -void QDM2Stream::process_synthesis_subpackets(QDM2SubPNode *list) { - struct QDM2SubPNode *nodes[4]; - - nodes[0] = qdm2_search_subpacket_type_in_list(list, 9); - if (nodes[0] != NULL) - process_subpacket_9(nodes[0]); - - nodes[1] = qdm2_search_subpacket_type_in_list(list, 10); - if (nodes[1] != NULL) - process_subpacket_10(nodes[1], nodes[1]->packet->size << 3); - else - process_subpacket_10(NULL, 0); - - nodes[2] = qdm2_search_subpacket_type_in_list(list, 11); - if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL) - process_subpacket_11(nodes[2], (nodes[2]->packet->size << 3)); - else - process_subpacket_11(NULL, 0); - - nodes[3] = qdm2_search_subpacket_type_in_list(list, 12); - if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL) - process_subpacket_12(nodes[3], (nodes[3]->packet->size << 3)); - else - process_subpacket_12(NULL, 0); -} - -/* - * Decode superblock, fill packet lists. - * - */ -void QDM2Stream::qdm2_decode_super_block(void) { - GetBitContext gb; - struct QDM2SubPacket header, *packet; - int i, packet_bytes, sub_packet_size, subPacketsD; - unsigned int next_index = 0; - - memset(_toneLevelIdxHi1, 0, sizeof(_toneLevelIdxHi1)); - memset(_toneLevelIdxMid, 0, sizeof(_toneLevelIdxMid)); - memset(_toneLevelIdxHi2, 0, sizeof(_toneLevelIdxHi2)); - - _subPacketsB = 0; - subPacketsD = 0; - - average_quantized_coeffs(); // average elements in quantized_coeffs[max_ch][10][8] - - initGetBits(&gb, _compressedData, _packetSize*8); - qdm2_decode_sub_packet_header(&gb, &header); - - if (header.type < 2 || header.type >= 8) { - _hasErrors = true; - error("QDM2 : bad superblock type"); - return; - } - - _superblocktype_2_3 = (header.type == 2 || header.type == 3); - packet_bytes = (_packetSize - getBitsCount(&gb) / 8); - - initGetBits(&gb, header.data, header.size*8); - - if (header.type == 2 || header.type == 4 || header.type == 5) { - int csum = 257 * getBits(&gb, 8) + 2 * getBits(&gb, 8); - - csum = qdm2_packet_checksum(_compressedData, _packetSize, csum); - - if (csum != 0) { - _hasErrors = true; - error("QDM2 : bad packet checksum"); - return; - } - } - - _subPacketListB[0].packet = NULL; - _subPacketListD[0].packet = NULL; - - for (i = 0; i < 6; i++) - if (--_fftLevelExp[i] < 0) - _fftLevelExp[i] = 0; - - for (i = 0; packet_bytes > 0; i++) { - int j; - - _subPacketListA[i].next = NULL; - - if (i > 0) { - _subPacketListA[i - 1].next = &_subPacketListA[i]; - - // seek to next block - initGetBits(&gb, header.data, header.size*8); - skipBits(&gb, next_index*8); - - if (next_index >= header.size) - break; - } - - // decode subpacket - packet = &_subPackets[i]; - qdm2_decode_sub_packet_header(&gb, packet); - next_index = packet->size + getBitsCount(&gb) / 8; - sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2; - - if (packet->type == 0) - break; - - if (sub_packet_size > packet_bytes) { - if (packet->type != 10 && packet->type != 11 && packet->type != 12) - break; - packet->size += packet_bytes - sub_packet_size; - } - - packet_bytes -= sub_packet_size; - - // add subpacket to 'all subpackets' list - _subPacketListA[i].packet = packet; - - // add subpacket to related list - if (packet->type == 8) { - error("Unsupported packet type 8"); - return; - } else if (packet->type >= 9 && packet->type <= 12) { - // packets for MPEG Audio like Synthesis Filter - QDM2_LIST_ADD(_subPacketListD, subPacketsD, packet); - } else if (packet->type == 13) { - for (j = 0; j < 6; j++) - _fftLevelExp[j] = getBits(&gb, 6); - } else if (packet->type == 14) { - for (j = 0; j < 6; j++) - _fftLevelExp[j] = qdm2_get_vlc(&gb, &_fftLevelExpVlc, 0, 2); - } else if (packet->type == 15) { - error("Unsupported packet type 15"); - return; - } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) { - // packets for FFT - QDM2_LIST_ADD(_subPacketListB, _subPacketsB, packet); - } - } // Packet bytes loop - -// **************************************************************** - if (_subPacketListD[0].packet != NULL) { - process_synthesis_subpackets(_subPacketListD); - _doSynthFilter = 1; - } else if (_doSynthFilter) { - process_subpacket_10(NULL, 0); - process_subpacket_11(NULL, 0); - process_subpacket_12(NULL, 0); - } -// **************************************************************** -} - -void QDM2Stream::qdm2_fft_init_coefficient(int sub_packet, int offset, int duration, - int channel, int exp, int phase) { - if (_fftCoefsMinIndex[duration] < 0) - _fftCoefsMinIndex[duration] = _fftCoefsIndex; - - _fftCoefs[_fftCoefsIndex].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet); - _fftCoefs[_fftCoefsIndex].channel = channel; - _fftCoefs[_fftCoefsIndex].offset = offset; - _fftCoefs[_fftCoefsIndex].exp = exp; - _fftCoefs[_fftCoefsIndex].phase = phase; - _fftCoefsIndex++; -} - -void QDM2Stream::qdm2_fft_decode_tones(int duration, GetBitContext *gb, int b) { - debug(1, "QDM2Stream::qdm2_fft_decode_tones() duration: %d b:%d", duration, b); - int channel, stereo, phase, exp; - int local_int_4, local_int_8, stereo_phase, local_int_10; - int local_int_14, stereo_exp, local_int_20, local_int_28; - int n, offset; - - local_int_4 = 0; - local_int_28 = 0; - local_int_20 = 2; - local_int_8 = (4 - duration); - local_int_10 = 1 << (_groupOrder - duration - 1); - offset = 1; - - while (1) { - if (_superblocktype_2_3) { - debug(1, "QDM2Stream::qdm2_fft_decode_tones() local_int_8: %d", local_int_8); - while ((n = qdm2_get_vlc(gb, &_vlcTabFftToneOffset[local_int_8], 1, 2)) < 2) { - debug(1, "QDM2Stream::qdm2_fft_decode_tones() local_int_8: %d", local_int_8); - offset = 1; - if (n == 0) { - local_int_4 += local_int_10; - local_int_28 += (1 << local_int_8); - } else { - local_int_4 += 8*local_int_10; - local_int_28 += (8 << local_int_8); - } - } - offset += (n - 2); - } else { - offset += qdm2_get_vlc(gb, &_vlcTabFftToneOffset[local_int_8], 1, 2); - while (offset >= (local_int_10 - 1)) { - offset += (1 - (local_int_10 - 1)); - local_int_4 += local_int_10; - local_int_28 += (1 << local_int_8); - } - } - - if (local_int_4 >= _blockSize) - return; - - local_int_14 = (offset >> local_int_8); - - if (_channels > 1) { - channel = getBits1(gb); - stereo = getBits1(gb); - } else { - channel = 0; - stereo = 0; - } - - exp = qdm2_get_vlc(gb, (b ? &_fftLevelExpVlc : &_fftLevelExpAltVlc), 0, 2); - exp += _fftLevelExp[fft_level_index_table[local_int_14]]; - exp = (exp < 0) ? 0 : exp; - - phase = getBits(gb, 3); - stereo_exp = 0; - stereo_phase = 0; - - if (stereo) { - stereo_exp = (exp - qdm2_get_vlc(gb, &_fftStereoExpVlc, 0, 1)); - stereo_phase = (phase - qdm2_get_vlc(gb, &_fftStereoPhaseVlc, 0, 1)); - if (stereo_phase < 0) - stereo_phase += 8; - } - - if (_frequencyRange > (local_int_14 + 1)) { - int sub_packet = (local_int_20 + local_int_28); - - qdm2_fft_init_coefficient(sub_packet, offset, duration, channel, exp, phase); - if (stereo) - qdm2_fft_init_coefficient(sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase); - } - - offset++; - } -} - -void QDM2Stream::qdm2_decode_fft_packets(void) { - debug(1, "QDM2Stream::qdm2_decode_fft_packets()"); - int i, j, min, max, value, type, unknown_flag; - GetBitContext gb; - - if (_subPacketListB[0].packet == NULL) - return; - - // reset minimum indexes for FFT coefficients - _fftCoefsIndex = 0; - for (i=0; i < 5; i++) - _fftCoefsMinIndex[i] = -1; - - // process subpackets ordered by type, largest type first - for (i = 0, max = 256; i < _subPacketsB; i++) { - QDM2SubPacket *packet= NULL; - - // find subpacket with largest type less than max - for (j = 0, min = 0; j < _subPacketsB; j++) { - value = _subPacketListB[j].packet->type; - if (value > min && value < max) { - min = value; - packet = _subPacketListB[j].packet; - } - } - - max = min; - - // check for errors (?) - if (!packet) - return; - - if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16])) - return; - - // decode FFT tones - debug(1, "QDM2Stream::qdm2_decode_fft_packets initGetBits() packet->size*8: %d", packet->size*8); - initGetBits(&gb, packet->data, packet->size*8); - - if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16]) - unknown_flag = 1; - else - unknown_flag = 0; - - type = packet->type; - - if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) { - int duration = _subSampling + 5 - (type & 15); - - if (duration >= 0 && duration < 4) { // TODO: Should be <= 4? - debug(1, "QDM2Stream::qdm2_decode_fft_packets qdm2_fft_decode_tones() #1"); - qdm2_fft_decode_tones(duration, &gb, unknown_flag); - } - } else if (type == 31) { - for (j=0; j < 4; j++) { - debug(1, "QDM2Stream::qdm2_decode_fft_packets qdm2_fft_decode_tones() #2"); - qdm2_fft_decode_tones(j, &gb, unknown_flag); - } - } else if (type == 46) { - for (j=0; j < 6; j++) - _fftLevelExp[j] = getBits(&gb, 6); - for (j=0; j < 4; j++) { - debug(1, "QDM2Stream::qdm2_decode_fft_packets qdm2_fft_decode_tones() #3"); - qdm2_fft_decode_tones(j, &gb, unknown_flag); - } - } - } // Loop on B packets - - // calculate maximum indexes for FFT coefficients - for (i = 0, j = -1; i < 5; i++) - if (_fftCoefsMinIndex[i] >= 0) { - if (j >= 0) - _fftCoefsMaxIndex[j] = _fftCoefsMinIndex[i]; - j = i; - } - if (j >= 0) - _fftCoefsMaxIndex[j] = _fftCoefsIndex; -} - -void QDM2Stream::qdm2_fft_generate_tone(FFTTone *tone) -{ - float level, f[6]; - int i; - QDM2Complex c; - const double iscale = 2.0 * PI / 512.0; - - tone->phase += tone->phase_shift; - - // calculate current level (maximum amplitude) of tone - level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level; - c.im = level * sin(tone->phase*iscale); - c.re = level * cos(tone->phase*iscale); - - // generate FFT coefficients for tone - if (tone->duration >= 3 || tone->cutoff >= 3) { - tone->complex[0].im += c.im; - tone->complex[0].re += c.re; - tone->complex[1].im -= c.im; - tone->complex[1].re -= c.re; - } else { - f[1] = -tone->table[4]; - f[0] = tone->table[3] - tone->table[0]; - f[2] = 1.0 - tone->table[2] - tone->table[3]; - f[3] = tone->table[1] + tone->table[4] - 1.0; - f[4] = tone->table[0] - tone->table[1]; - f[5] = tone->table[2]; - for (i = 0; i < 2; i++) { - tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i]; - tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]); - } - for (i = 0; i < 4; i++) { - tone->complex[i].re += c.re * f[i+2]; - tone->complex[i].im += c.im * f[i+2]; - } - } - - // copy the tone if it has not yet died out - if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) { - memcpy(&_fftTones[_fftToneEnd], tone, sizeof(FFTTone)); - _fftToneEnd = (_fftToneEnd + 1) % 1000; - } -} - -void QDM2Stream::qdm2_fft_tone_synthesizer(uint8 sub_packet) { - int i, j, ch; - const double iscale = 0.25 * PI; - - for (ch = 0; ch < _channels; ch++) { - memset(_fft.complex[ch], 0, _frameSize * sizeof(QDM2Complex)); - } - - // apply FFT tones with duration 4 (1 FFT period) - if (_fftCoefsMinIndex[4] >= 0) - for (i = _fftCoefsMinIndex[4]; i < _fftCoefsMaxIndex[4]; i++) { - float level; - QDM2Complex c; - - if (_fftCoefs[i].sub_packet != sub_packet) - break; - - ch = (_channels == 1) ? 0 : _fftCoefs[i].channel; - level = (_fftCoefs[i].exp < 0) ? 0.0 : fft_tone_level_table[_superblocktype_2_3 ? 0 : 1][_fftCoefs[i].exp & 63]; - - c.re = level * cos(_fftCoefs[i].phase * iscale); - c.im = level * sin(_fftCoefs[i].phase * iscale); - _fft.complex[ch][_fftCoefs[i].offset + 0].re += c.re; - _fft.complex[ch][_fftCoefs[i].offset + 0].im += c.im; - _fft.complex[ch][_fftCoefs[i].offset + 1].re -= c.re; - _fft.complex[ch][_fftCoefs[i].offset + 1].im -= c.im; - } - - // generate existing FFT tones - for (i = _fftToneEnd; i != _fftToneStart; ) { - qdm2_fft_generate_tone(&_fftTones[_fftToneStart]); - _fftToneStart = (_fftToneStart + 1) % 1000; - } - - // create and generate new FFT tones with duration 0 (long) to 3 (short) - for (i = 0; i < 4; i++) - if (_fftCoefsMinIndex[i] >= 0) { - for (j = _fftCoefsMinIndex[i]; j < _fftCoefsMaxIndex[i]; j++) { - int offset, four_i; - FFTTone tone; - - if (_fftCoefs[j].sub_packet != sub_packet) - break; - - four_i = (4 - i); - offset = _fftCoefs[j].offset >> four_i; - ch = (_channels == 1) ? 0 : _fftCoefs[j].channel; - - if (offset < _frequencyRange) { - if (offset < 2) - tone.cutoff = offset; - else - tone.cutoff = (offset >= 60) ? 3 : 2; - - tone.level = (_fftCoefs[j].exp < 0) ? 0.0 : fft_tone_level_table[_superblocktype_2_3 ? 0 : 1][_fftCoefs[j].exp & 63]; - tone.complex = &_fft.complex[ch][offset]; - tone.table = fft_tone_sample_table[i][_fftCoefs[j].offset - (offset << four_i)]; - tone.phase = 64 * _fftCoefs[j].phase - (offset << 8) - 128; - tone.phase_shift = (2 * _fftCoefs[j].offset + 1) << (7 - four_i); - tone.duration = i; - tone.time_index = 0; - - qdm2_fft_generate_tone(&tone); - } - } - _fftCoefsMinIndex[i] = j; - } -} - -void QDM2Stream::qdm2_calculate_fft(int channel) { - debug(1, "QDM2Stream::qdm2_calculate_fft channel: %d", channel); - const float gain = (_channels == 1 && _channels == 2) ? 0.5f : 1.0f; - int i; - - _fft.complex[channel][0].re *= 2.0f; - _fft.complex[channel][0].im = 0.0f; - - //debug(1, "QDM2Stream::qdm2_calculate_fft _fft.complex[channel][0].re: %lf", _fft.complex[channel][0].re); - //debug(1, "QDM2Stream::qdm2_calculate_fft _fft.complex[channel][0].im: %lf", _fft.complex[channel][0].im); - - rdftCalc(&_rdftCtx, (float *)_fft.complex[channel]); - - // add samples to output buffer - for (i = 0; i < ((_fftFrameSize + 15) & ~15); i++) - _outputBuffer[_channels * i + channel] += ((float *) _fft.complex[channel])[i] * gain; -} - -/** - * @param index subpacket number - */ -void QDM2Stream::qdm2_synthesis_filter(uint8 index) -{ - int16 samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE]; - int i, k, ch, sb_used, sub_sampling, dither_state = 0; - - // copy sb_samples - sb_used = QDM2_SB_USED(_subSampling); - - for (ch = 0; ch < _channels; ch++) - for (i = 0; i < 8; i++) - for (k = sb_used; k < 32; k++) - _sbSamples[ch][(8 * index) + i][k] = 0; - - for (ch = 0; ch < _channels; ch++) { - int16 *samples_ptr = samples + ch; - - for (i = 0; i < 8; i++) { - ff_mpa_synth_filter(_synthBuf[ch], &(_synthBufOffset[ch]), - ff_mpa_synth_window, &dither_state, - samples_ptr, _channels, - _sbSamples[ch][(8 * index) + i]); - samples_ptr += 32 * _channels; - } - } - - // add samples to output buffer - sub_sampling = (4 >> _subSampling); - - for (ch = 0; ch < _channels; ch++) - for (i = 0; i < _sFrameSize; i++) - _outputBuffer[_channels * i + ch] += (float)(samples[_channels * sub_sampling * i + ch] >> (sizeof(int16)*8-16)); -} - -int QDM2Stream::qdm2_decodeFrame(Common::SeekableReadStream *in) { - debug(1, "QDM2Stream::qdm2_decodeFrame in->pos(): %d in->size(): %d", in->pos(), in->size()); - int ch, i; - const int frame_size = (_sFrameSize * _channels); - - // select input buffer - if(in->eos() || in->size() == in->pos()) { - debug(1, "QDM2Stream::qdm2_decodeFrame End of Input Stream"); - return 0; - } - if((in->size() - in->pos()) < _packetSize) { - debug(1, "QDM2Stream::qdm2_decodeFrame Insufficient Packet Data in Input Stream Found: %d Need: %d", in->size() - in->pos(), _packetSize); - return 0; - } - - in->read(_compressedData, _packetSize); - debug(1, "QDM2Stream::qdm2_decodeFrame constructed input data"); - - // copy old block, clear new block of output samples - memmove(_outputBuffer, &_outputBuffer[frame_size], frame_size * sizeof(float)); - memset(&_outputBuffer[frame_size], 0, frame_size * sizeof(float)); - debug(1, "QDM2Stream::qdm2_decodeFrame cleared outputBuffer"); - - // decode block of QDM2 compressed data - debug(1, "QDM2Stream::qdm2_decodeFrame decode block of QDM2 compressed data"); - if (_subPacket == 0) { - _hasErrors = false; // reset it for a new super block - debug(1, "QDM2 : Superblock follows"); - qdm2_decode_super_block(); - } - - // parse subpackets - debug(1, "QDM2Stream::qdm2_decodeFrame parse subpackets"); - if (!_hasErrors) { - if (_subPacket == 2) { - debug(1, "QDM2Stream::qdm2_decodeFrame qdm2_decode_fft_packets()"); - qdm2_decode_fft_packets(); - } - - debug(1, "QDM2Stream::qdm2_decodeFrame qdm2_fft_tone_synthesizer(%d)", _subPacket); - qdm2_fft_tone_synthesizer(_subPacket); - } - - // sound synthesis stage 1 (FFT) - debug(1, "QDM2Stream::qdm2_decodeFrame sound synthesis stage 1 (FFT)"); - for (ch = 0; ch < _channels; ch++) { - qdm2_calculate_fft(ch); - - if (!_hasErrors && _subPacketListC[0].packet != NULL) { - error("QDM2 : has errors, and C list is not empty"); - return 0; - } - } - - // sound synthesis stage 2 (MPEG audio like synthesis filter) - debug(1, "QDM2Stream::qdm2_decodeFrame sound synthesis stage 2 (MPEG audio like synthesis filter)"); - if (!_hasErrors && _doSynthFilter) - qdm2_synthesis_filter(_subPacket); - - _subPacket = (_subPacket + 1) % 16; - - if(_hasErrors) - warning("QDM2 Packet error..."); - - // clip and convert output float[] to 16bit signed samples - debug(1, "QDM2Stream::qdm2_decodeFrame clip and convert output float[] to 16bit signed samples"); - -/* - debugN(1, "Input Data Packet:"); - for(i = 0; i < _packetSize; i++) { - debugN(1, " %d", _compressedData[i]); - } - debugN(1, " Output Data Packet:"); - for(i = 0; i < frame_size; i++) { - debugN(1, " %d", (int)_outputBuffer[i]); - } - debug(1, ""); -*/ - - for (i = 0; i < frame_size; i++) { - //debug(1, "QDM2Stream::qdm2_decodeFrame i: %d", i); - int value = (int)_outputBuffer[i]; - - if (value > SOFTCLIP_THRESHOLD) - value = (value > HARDCLIP_THRESHOLD) ? 32767 : _softclipTable[ value - SOFTCLIP_THRESHOLD]; - else if (value < -SOFTCLIP_THRESHOLD) - value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -_softclipTable[-value - SOFTCLIP_THRESHOLD]; - - _outputSamples.push_back(value); - } - return frame_size; -} - -int QDM2Stream::readBuffer(int16 *buffer, const int numSamples) { - debug(1, "QDM2Stream::readBuffer numSamples: %d", numSamples); - int32 decodedSamples = _outputSamples.size(); - int32 i; - - //while((int)_outputSamples.size() < numSamples) { - while(!_stream->eos() && _stream->pos() != _stream->size()) { - i = qdm2_decodeFrame(_stream); - if(i == 0) - break; // Out Of Decode Frames... - decodedSamples += i; - } - if(decodedSamples > numSamples) - decodedSamples = numSamples; - - for(i = 0; i < decodedSamples; i++) - buffer[i] = _outputSamples.remove_at(0); - - return decodedSamples; -} - -AudioStream *makeQDM2Stream(Common::SeekableReadStream *stream, Common::SeekableReadStream *extraData) { - return new QDM2Stream(stream, extraData); -} - -} // End of namespace Audio - -#endif diff --git a/sound/decoders/qdm2.h b/sound/decoders/qdm2.h deleted file mode 100644 index 842ede3de0..0000000000 --- a/sound/decoders/qdm2.h +++ /dev/null @@ -1,51 +0,0 @@ -/* 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 - -namespace Common { - class SeekableReadStream; -} - -namespace Audio { - class AudioStream; - -/** - * Create a new AudioStream from the QDM2 data in the given stream. - * - * @param stream the SeekableReadStream from which to read the FLAC data - * @param extraData the QuickTime extra data stream - * @return a new AudioStream, or NULL, if an error occured - */ -AudioStream *makeQDM2Stream(Common::SeekableReadStream *stream, Common::SeekableReadStream *extraData); - -} // End of namespace Audio - -#endif // SOUND_QDM2_H -#endif // Mohawk/Plugins guard diff --git a/sound/decoders/qdm2data.h b/sound/decoders/qdm2data.h deleted file mode 100644 index 4c13328dd6..0000000000 --- a/sound/decoders/qdm2data.h +++ /dev/null @@ -1,531 +0,0 @@ -/* ScummVM - Graphic Adventure Engine - * - * ScummVM is the legal property of its developers, whose names - * are too numerous to list here. Please refer to the COPYRIGHT - * file distributed with this source distribution. - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * as published by the Free Software Foundation; either version 2 - * of the License, or (at your option) any later version. - - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA. - * - * $URL$ - * $Id$ - * - */ - -#ifndef SOUND_QDM2DATA_H -#define SOUND_QDM2DATA_H - -#include "common/scummsys.h" - -namespace Audio { - -/// VLC TABLES - -// values in this table range from -1..23; adjust retrieved value by -1 -static const uint16 vlc_tab_level_huffcodes[24] = { - 0x037c, 0x0004, 0x003c, 0x004c, 0x003a, 0x002c, 0x001c, 0x001a, - 0x0024, 0x0014, 0x0001, 0x0002, 0x0000, 0x0003, 0x0007, 0x0005, - 0x0006, 0x0008, 0x0009, 0x000a, 0x000c, 0x00fc, 0x007c, 0x017c -}; - -static const byte vlc_tab_level_huffbits[24] = { - 10, 6, 7, 7, 6, 6, 6, 6, 6, 5, 4, 4, 4, 3, 3, 3, 3, 4, 4, 5, 7, 8, 9, 10 -}; - -// values in this table range from -1..36; adjust retrieved value by -1 -static const uint16 vlc_tab_diff_huffcodes[37] = { - 0x1c57, 0x0004, 0x0000, 0x0001, 0x0003, 0x0002, 0x000f, 0x000e, - 0x0007, 0x0016, 0x0037, 0x0027, 0x0026, 0x0066, 0x0006, 0x0097, - 0x0046, 0x01c6, 0x0017, 0x0786, 0x0086, 0x0257, 0x00d7, 0x0357, - 0x00c6, 0x0386, 0x0186, 0x0000, 0x0157, 0x0c57, 0x0057, 0x0000, - 0x0b86, 0x0000, 0x1457, 0x0000, 0x0457 -}; - -static const byte vlc_tab_diff_huffbits[37] = { - 13, 3, 3, 2, 3, 3, 4, 4, 6, 5, 6, 6, 7, 7, 8, 8, - 8, 9, 8, 11, 9, 10, 8, 10, 9, 12, 10, 0, 10, 13, 11, 0, - 12, 0, 13, 0, 13 -}; - -// values in this table range from -1..5; adjust retrieved value by -1 -static const byte vlc_tab_run_huffcodes[6] = { - 0x1f, 0x00, 0x01, 0x03, 0x07, 0x0f -}; - -static const byte vlc_tab_run_huffbits[6] = { - 5, 1, 2, 3, 4, 5 -}; - -// values in this table range from -1..19; adjust retrieved value by -1 -static const uint16 vlc_tab_tone_level_idx_hi1_huffcodes[20] = { - 0x5714, 0x000c, 0x0002, 0x0001, 0x0000, 0x0004, 0x0034, 0x0054, - 0x0094, 0x0014, 0x0114, 0x0214, 0x0314, 0x0614, 0x0e14, 0x0f14, - 0x2714, 0x0714, 0x1714, 0x3714 -}; - -static const byte vlc_tab_tone_level_idx_hi1_huffbits[20] = { - 15, 4, 2, 1, 3, 5, 6, 7, 8, 10, 10, 11, 11, 12, 12, 12, 14, 14, 15, 14 -}; - -// values in this table range from -1..23; adjust retrieved value by -1 -static const uint16 vlc_tab_tone_level_idx_mid_huffcodes[24] = { - 0x0fea, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, 0x03ea, 0x00ea, 0x002a, 0x001a, - 0x0006, 0x0001, 0x0000, 0x0002, 0x000a, 0x006a, 0x01ea, 0x07ea -}; - -static const byte vlc_tab_tone_level_idx_mid_huffbits[24] = { - 12, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 11, 9, 7, 5, 3, 1, 2, 4, 6, 8, 10, 12 -}; - -// values in this table range from -1..23; adjust retrieved value by -1 -static const uint16 vlc_tab_tone_level_idx_hi2_huffcodes[24] = { - 0x0664, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0064, 0x00e4, - 0x00a4, 0x0068, 0x0004, 0x0008, 0x0014, 0x0018, 0x0000, 0x0001, - 0x0002, 0x0003, 0x000c, 0x0028, 0x0024, 0x0164, 0x0000, 0x0264 -}; - -static const byte vlc_tab_tone_level_idx_hi2_huffbits[24] = { - 11, 0, 0, 0, 0, 0, 10, 8, 8, 7, 6, 6, 5, 5, 4, 2, 2, 2, 4, 7, 8, 9, 0, 11 -}; - -// values in this table range from -1..8; adjust retrieved value by -1 -static const byte vlc_tab_type30_huffcodes[9] = { - 0x3c, 0x06, 0x00, 0x01, 0x03, 0x02, 0x04, 0x0c, 0x1c -}; - -static const byte vlc_tab_type30_huffbits[9] = { - 6, 3, 3, 2, 2, 3, 4, 5, 6 -}; - -// values in this table range from -1..9; adjust retrieved value by -1 -static const byte vlc_tab_type34_huffcodes[10] = { - 0x18, 0x00, 0x01, 0x04, 0x05, 0x07, 0x03, 0x02, 0x06, 0x08 -}; - -static const byte vlc_tab_type34_huffbits[10] = { - 5, 4, 3, 3, 3, 3, 3, 3, 3, 5 -}; - -// values in this table range from -1..22; adjust retrieved value by -1 -static const uint16 vlc_tab_fft_tone_offset_0_huffcodes[23] = { - 0x038e, 0x0001, 0x0000, 0x0022, 0x000a, 0x0006, 0x0012, 0x0002, - 0x001e, 0x003e, 0x0056, 0x0016, 0x000e, 0x0032, 0x0072, 0x0042, - 0x008e, 0x004e, 0x00f2, 0x002e, 0x0036, 0x00c2, 0x018e -}; - -static const byte vlc_tab_fft_tone_offset_0_huffbits[23] = { - 10, 1, 2, 6, 4, 5, 6, 7, 6, 6, 7, 7, 8, 7, 8, 8, 9, 7, 8, 6, 6, 8, 10 -}; - -// values in this table range from -1..27; adjust retrieved value by -1 -static const uint16 vlc_tab_fft_tone_offset_1_huffcodes[28] = { - 0x07a4, 0x0001, 0x0020, 0x0012, 0x001c, 0x0008, 0x0006, 0x0010, - 0x0000, 0x0014, 0x0004, 0x0032, 0x0070, 0x000c, 0x0002, 0x003a, - 0x001a, 0x002c, 0x002a, 0x0022, 0x0024, 0x000a, 0x0064, 0x0030, - 0x0062, 0x00a4, 0x01a4, 0x03a4 -}; - -static const byte vlc_tab_fft_tone_offset_1_huffbits[28] = { - 11, 1, 6, 6, 5, 4, 3, 6, 6, 5, 6, 6, 7, 6, 6, 6, - 6, 6, 6, 7, 8, 6, 7, 7, 7, 9, 10, 11 -}; - -// values in this table range from -1..31; adjust retrieved value by -1 -static const uint16 vlc_tab_fft_tone_offset_2_huffcodes[32] = { - 0x1760, 0x0001, 0x0000, 0x0082, 0x000c, 0x0006, 0x0003, 0x0007, - 0x0008, 0x0004, 0x0010, 0x0012, 0x0022, 0x001a, 0x0000, 0x0020, - 0x000a, 0x0040, 0x004a, 0x006a, 0x002a, 0x0042, 0x0002, 0x0060, - 0x00aa, 0x00e0, 0x00c2, 0x01c2, 0x0160, 0x0360, 0x0760, 0x0f60 -}; - -static const byte vlc_tab_fft_tone_offset_2_huffbits[32] = { - 13, 2, 0, 8, 4, 3, 3, 3, 4, 4, 5, 5, 6, 5, 7, 7, - 7, 7, 7, 7, 8, 8, 8, 9, 8, 8, 9, 9, 10, 11, 13, 12 -}; - -// values in this table range from -1..34; adjust retrieved value by -1 -static const uint16 vlc_tab_fft_tone_offset_3_huffcodes[35] = { - 0x33ea, 0x0005, 0x0000, 0x000c, 0x0000, 0x0006, 0x0003, 0x0008, - 0x0002, 0x0001, 0x0004, 0x0007, 0x001a, 0x000f, 0x001c, 0x002c, - 0x000a, 0x001d, 0x002d, 0x002a, 0x000d, 0x004c, 0x008c, 0x006a, - 0x00cd, 0x004d, 0x00ea, 0x020c, 0x030c, 0x010c, 0x01ea, 0x07ea, - 0x0bea, 0x03ea, 0x13ea -}; - -static const byte vlc_tab_fft_tone_offset_3_huffbits[35] = { - 14, 4, 0, 10, 4, 3, 3, 4, 4, 3, 4, 4, 5, 4, 5, 6, - 6, 5, 6, 7, 7, 7, 8, 8, 8, 8, 9, 10, 10, 10, 10, 11, - 12, 13, 14 -}; - -// values in this table range from -1..37; adjust retrieved value by -1 -static const uint16 vlc_tab_fft_tone_offset_4_huffcodes[38] = { - 0x5282, 0x0016, 0x0000, 0x0136, 0x0004, 0x0000, 0x0007, 0x000a, - 0x000e, 0x0003, 0x0001, 0x000d, 0x0006, 0x0009, 0x0012, 0x0005, - 0x0025, 0x0022, 0x0015, 0x0002, 0x0076, 0x0035, 0x0042, 0x00c2, - 0x0182, 0x00b6, 0x0036, 0x03c2, 0x0482, 0x01c2, 0x0682, 0x0882, - 0x0a82, 0x0082, 0x0282, 0x1282, 0x3282, 0x2282 -}; - -static const byte vlc_tab_fft_tone_offset_4_huffbits[38] = { - 15, 6, 0, 9, 3, 3, 3, 4, 4, 3, 4, 4, 5, 4, 5, 6, - 6, 6, 6, 8, 7, 6, 8, 9, 9, 8, 9, 10, 11, 10, 11, 12, - 12, 12, 14, 15, 14, 14 -}; - -/// FFT TABLES - -// values in this table range from -1..27; adjust retrieved value by -1 -static const uint16 fft_level_exp_alt_huffcodes[28] = { - 0x1ec6, 0x0006, 0x00c2, 0x0142, 0x0242, 0x0246, 0x00c6, 0x0046, - 0x0042, 0x0146, 0x00a2, 0x0062, 0x0026, 0x0016, 0x000e, 0x0005, - 0x0004, 0x0003, 0x0000, 0x0001, 0x000a, 0x0012, 0x0002, 0x0022, - 0x01c6, 0x02c6, 0x06c6, 0x0ec6 -}; - -static const byte fft_level_exp_alt_huffbits[28] = { - 13, 7, 8, 9, 10, 10, 10, 10, 10, 9, 8, 7, 6, 5, 4, 3, - 3, 2, 3, 3, 4, 5, 7, 8, 9, 11, 12, 13 -}; - -// values in this table range from -1..19; adjust retrieved value by -1 -static const uint16 fft_level_exp_huffcodes[20] = { - 0x0f24, 0x0001, 0x0002, 0x0000, 0x0006, 0x0005, 0x0007, 0x000c, - 0x000b, 0x0014, 0x0013, 0x0004, 0x0003, 0x0023, 0x0064, 0x00a4, - 0x0024, 0x0124, 0x0324, 0x0724 -}; - -static const byte fft_level_exp_huffbits[20] = { - 12, 3, 3, 3, 3, 3, 3, 4, 4, 5, 5, 6, 6, 6, 7, 8, 9, 10, 11, 12 -}; - -// values in this table range from -1..6; adjust retrieved value by -1 -static const byte fft_stereo_exp_huffcodes[7] = { - 0x3e, 0x01, 0x00, 0x02, 0x06, 0x0e, 0x1e -}; - -static const byte fft_stereo_exp_huffbits[7] = { - 6, 1, 2, 3, 4, 5, 6 -}; - -// values in this table range from -1..8; adjust retrieved value by -1 -static const byte fft_stereo_phase_huffcodes[9] = { - 0x35, 0x02, 0x00, 0x01, 0x0d, 0x15, 0x05, 0x09, 0x03 -}; - -static const byte fft_stereo_phase_huffbits[9] = { - 6, 2, 2, 4, 4, 6, 5, 4, 2 -}; - -static const int fft_cutoff_index_table[4][2] = { - { 1, 2 }, {-1, 0 }, {-1,-2 }, { 0, 0 } -}; - -static const int16 fft_level_index_table[256] = { - 0, 0, 0, 0, 0, 0, 0, 0, 1, 1, 1, 1, 1, 1, 1, 1, - 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, - 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, - 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, - 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, - 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, - 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, - 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, - 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, - 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, - 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, - 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, - 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, - 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, - 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, - 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, -}; - -static const byte last_coeff[3] = { - 4, 7, 10 -}; - -static const byte coeff_per_sb_for_avg[3][30] = { - { 0, 1, 1, 1, 1, 2, 2, 2, 2, 2, 2, 2, 2, 2, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3 }, - { 0, 1, 2, 2, 3, 3, 4, 4, 4, 4, 4, 4, 5, 5, 5, 5, 5, 5, 5, 5, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6 }, - { 0, 1, 2, 3, 4, 4, 5, 5, 6, 6, 6, 6, 7, 7, 7, 7, 8, 8, 8, 8, 8, 8, 9, 9, 9, 9, 9, 9, 9, 9 } -}; - -static const uint32 dequant_table[3][10][30] = { - { { 256, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, - { 0, 256, 256, 205, 154, 102, 51, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, - { 0, 0, 0, 51, 102, 154, 205, 256, 238, 219, 201, 183, 165, 146, 128, 110, 91, 73, 55, 37, 18, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, - { 0, 0, 0, 0, 0, 0, 0, 0, 18, 37, 55, 73, 91, 110, 128, 146, 165, 183, 201, 219, 238, 256, 228, 199, 171, 142, 114, 85, 57, 28 }, - { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, - { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, - { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, - { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, - { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, - { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 } }, - { { 256, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, - { 0, 256, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, - { 0, 0, 256, 171, 85, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, - { 0, 0, 0, 85, 171, 256, 171, 85, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, - { 0, 0, 0, 0, 0, 0, 85, 171, 256, 219, 183, 146, 110, 73, 37, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, - { 0, 0, 0, 0, 0, 0, 0, 0, 0, 37, 73, 110, 146, 183, 219, 256, 228, 199, 171, 142, 114, 85, 57, 28, 0, 0, 0, 0, 0, 0 }, - { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 28, 57, 85, 114, 142, 171, 199, 228, 256, 213, 171, 128, 85, 43 }, - { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, - { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, - { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 } }, - { { 256, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, - { 0, 256, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, - { 0, 0, 256, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, - { 0, 0, 0, 256, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, - { 0, 0, 0, 0, 256, 256, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, - { 0, 0, 0, 0, 0, 0, 256, 171, 85, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, - { 0, 0, 0, 0, 0, 0, 0, 85, 171, 256, 192, 128, 64, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, - { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 64, 128, 192, 256, 205, 154, 102, 51, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, - { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 51, 102, 154, 205, 256, 213, 171, 128, 85, 43, 0, 0, 0, 0, 0, 0 }, - { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 43, 85, 128, 171, 213, 256, 213, 171, 128, 85, 43 } } -}; - -static const byte coeff_per_sb_for_dequant[3][30] = { - { 0, 1, 1, 1, 1, 1, 1, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 3, 3, 3, 3, 3, 3, 3, 3, 3 }, - { 0, 1, 2, 2, 2, 3, 3, 3, 4, 4, 4, 4, 4, 4, 4, 5, 5, 5, 5, 5, 5, 5, 5, 5, 6, 6, 6, 6, 6, 6 }, - { 0, 1, 2, 3, 4, 4, 5, 5, 5, 6, 6, 6, 6, 7, 7, 7, 7, 7, 8, 8, 8, 8, 8, 8, 9, 9, 9, 9, 9, 9 } -}; - -// first index is subband, 2nd index is 0, 1 or 3 (2 is unused) -static const int8 tone_level_idx_offset_table[30][4] = { - { -50, -50, 0, -50 }, - { -50, -50, 0, -50 }, - { -50, -9, 0, -19 }, - { -16, -6, 0, -12 }, - { -11, -4, 0, -8 }, - { -8, -3, 0, -6 }, - { -7, -3, 0, -5 }, - { -6, -2, 0, -4 }, - { -5, -2, 0, -3 }, - { -4, -1, 0, -3 }, - { -4, -1, 0, -2 }, - { -3, -1, 0, -2 }, - { -3, -1, 0, -2 }, - { -3, -1, 0, -2 }, - { -2, -1, 0, -1 }, - { -2, -1, 0, -1 }, - { -2, -1, 0, -1 }, - { -2, 0, 0, -1 }, - { -2, 0, 0, -1 }, - { -1, 0, 0, -1 }, - { -1, 0, 0, -1 }, - { -1, 0, 0, -1 }, - { -1, 0, 0, -1 }, - { -1, 0, 0, -1 }, - { -1, 0, 0, -1 }, - { -1, 0, 0, -1 }, - { -1, 0, 0, 0 }, - { -1, 0, 0, 0 }, - { -1, 0, 0, 0 }, - { -1, 0, 0, 0 } -}; - -/* all my samples have 1st index 0 or 1 */ -/* second index is subband, only indexes 0-29 seem to be used */ -static const int8 coding_method_table[5][30] = { - { 34, 30, 24, 24, 16, 16, 16, 16, 10, 10, 10, 10, 10, 10, 10, - 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10 - }, - { 34, 30, 24, 24, 16, 16, 16, 16, 10, 10, 10, 10, 10, 10, 10, - 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10 - }, - { 34, 30, 30, 30, 24, 24, 16, 16, 16, 16, 16, 16, 10, 10, 10, - 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10 - }, - { 34, 34, 30, 30, 24, 24, 24, 24, 16, 16, 16, 16, 16, 16, 16, - 16, 16, 16, 16, 16, 16, 16, 10, 10, 10, 10, 10, 10, 10, 10 - }, - { 34, 34, 30, 30, 30, 30, 30, 30, 24, 24, 24, 24, 24, 24, 24, - 24, 24, 24, 24, 24, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16 - }, -}; - -static const int vlc_stage3_values[60] = { - 0, 1, 2, 3, 4, 6, 8, 10, 12, 16, 20, 24, - 28, 36, 44, 52, 60, 76, 92, 108, 124, 156, 188, 220, - 252, 316, 380, 444, 508, 636, 764, 892, 1020, 1276, 1532, 1788, - 2044, 2556, 3068, 3580, 4092, 5116, 6140, 7164, 8188, 10236, 12284, 14332, - 16380, 20476, 24572, 28668, 32764, 40956, 49148, 57340, 65532, 81916, 98300,114684 -}; - -static const float fft_tone_sample_table[4][16][5] = { - { { .0100000000f,-.0037037037f,-.0020000000f,-.0069444444f,-.0018416207f }, - { .0416666667f, .0000000000f, .0000000000f,-.0208333333f,-.0123456791f }, - { .1250000000f, .0558035709f, .0330687836f,-.0164473690f,-.0097465888f }, - { .1562500000f, .0625000000f, .0370370370f,-.0062500000f,-.0037037037f }, - { .1996007860f, .0781250000f, .0462962948f, .0022727272f, .0013468013f }, - { .2000000000f, .0625000000f, .0370370373f, .0208333333f, .0074074073f }, - { .2127659619f, .0555555556f, .0329218097f, .0208333333f, .0123456791f }, - { .2173913121f, .0473484844f, .0280583613f, .0347222239f, .0205761325f }, - { .2173913121f, .0347222239f, .0205761325f, .0473484844f, .0280583613f }, - { .2127659619f, .0208333333f, .0123456791f, .0555555556f, .0329218097f }, - { .2000000000f, .0208333333f, .0074074073f, .0625000000f, .0370370370f }, - { .1996007860f, .0022727272f, .0013468013f, .0781250000f, .0462962948f }, - { .1562500000f,-.0062500000f,-.0037037037f, .0625000000f, .0370370370f }, - { .1250000000f,-.0164473690f,-.0097465888f, .0558035709f, .0330687836f }, - { .0416666667f,-.0208333333f,-.0123456791f, .0000000000f, .0000000000f }, - { .0100000000f,-.0069444444f,-.0018416207f,-.0037037037f,-.0020000000f } }, - - { { .0050000000f,-.0200000000f, .0125000000f,-.3030303030f, .0020000000f }, - { .1041666642f, .0400000000f,-.0250000000f, .0333333333f,-.0200000000f }, - { .1250000000f, .0100000000f, .0142857144f,-.0500000007f,-.0200000000f }, - { .1562500000f,-.0006250000f,-.00049382716f,-.000625000f,-.00049382716f }, - { .1562500000f,-.0006250000f,-.00049382716f,-.000625000f,-.00049382716f }, - { .1250000000f,-.0500000000f,-.0200000000f, .0100000000f, .0142857144f }, - { .1041666667f, .0333333333f,-.0200000000f, .0400000000f,-.0250000000f }, - { .0050000000f,-.3030303030f, .0020000001f,-.0200000000f, .0125000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f } }, - - { { .1428571492f, .1250000000f,-.0285714287f,-.0357142873f, .0208333333f }, - { .1818181818f, .0588235296f, .0333333333f, .0212765951f, .0100000000f }, - { .1818181818f, .0212765951f, .0100000000f, .0588235296f, .0333333333f }, - { .1428571492f,-.0357142873f, .0208333333f, .1250000000f,-.0285714287f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f } }, - - { { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, - { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f } } -}; - -static const float fft_tone_level_table[2][64] = { { -// pow ~ (i > 46) ? 0 : (((((i & 1) ? 431 : 304) << (i >> 1))) / 1024.0); - 0.17677669f, 0.42677650f, 0.60355347f, 0.85355347f, - 1.20710683f, 1.68359375f, 2.37500000f, 3.36718750f, - 4.75000000f, 6.73437500f, 9.50000000f, 13.4687500f, - 19.0000000f, 26.9375000f, 38.0000000f, 53.8750000f, - 76.0000000f, 107.750000f, 152.000000f, 215.500000f, - 304.000000f, 431.000000f, 608.000000f, 862.000000f, - 1216.00000f, 1724.00000f, 2432.00000f, 3448.00000f, - 4864.00000f, 6896.00000f, 9728.00000f, 13792.0000f, - 19456.0000f, 27584.0000f, 38912.0000f, 55168.0000f, - 77824.0000f, 110336.000f, 155648.000f, 220672.000f, - 311296.000f, 441344.000f, 622592.000f, 882688.000f, - 1245184.00f, 1765376.00f, 2490368.00f, 0.00000000f, - 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f, - 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f, - 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f, - 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f, - }, { -// pow = (i > 45) ? 0 : ((((i & 1) ? 431 : 304) << (i >> 1)) / 512.0); - 0.59375000f, 0.84179688f, 1.18750000f, 1.68359375f, - 2.37500000f, 3.36718750f, 4.75000000f, 6.73437500f, - 9.50000000f, 13.4687500f, 19.0000000f, 26.9375000f, - 38.0000000f, 53.8750000f, 76.0000000f, 107.750000f, - 152.000000f, 215.500000f, 304.000000f, 431.000000f, - 608.000000f, 862.000000f, 1216.00000f, 1724.00000f, - 2432.00000f, 3448.00000f, 4864.00000f, 6896.00000f, - 9728.00000f, 13792.0000f, 19456.0000f, 27584.0000f, - 38912.0000f, 55168.0000f, 77824.0000f, 110336.000f, - 155648.000f, 220672.000f, 311296.000f, 441344.000f, - 622592.000f, 882688.000f, 1245184.00f, 1765376.00f, - 2490368.00f, 3530752.00f, 0.00000000f, 0.00000000f, - 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f, - 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f, - 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f, - 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f -} }; - -static const float fft_tone_envelope_table[4][31] = { - { .009607375f, .038060248f, .084265202f, .146446645f, .222214907f, .308658302f, - .402454883f, .500000060f, .597545207f, .691341758f, .777785182f, .853553414f, - .915734828f, .961939812f, .990392685f, 1.00000000f, .990392625f, .961939752f, - .915734768f, .853553295f, .777785063f, .691341639f, .597545087f, .500000000f, - .402454853f, .308658272f, .222214878f, .146446615f, .084265172f, .038060218f, - .009607345f }, - { .038060248f, .146446645f, .308658302f, .500000060f, .691341758f, .853553414f, - .961939812f, 1.00000000f, .961939752f, .853553295f, .691341639f, .500000000f, - .308658272f, .146446615f, .038060218f, .000000000f, .000000000f, .000000000f, - .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, - .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, - .000000000f }, - { .146446645f, .500000060f, .853553414f, 1.00000000f, .853553295f, .500000000f, - .146446615f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, - .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, - .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, - .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, - .000000000f }, - { .500000060f, 1.00000000f, .500000000f, .000000000f, .000000000f, .000000000f, - .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, - .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, - .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, - .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, - .000000000f } -}; - -static const float sb_noise_attenuation[32] = { - 0.0f, 0.0f, 0.3f, 0.4f, 0.5f, 0.7f, 1.0f, 1.0f, - 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, - 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, - 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, -}; - -static const byte fft_subpackets[32] = { - 0, 0, 0, 0, 0, 0, 0, 0, 1, 1, 1, 1, 1, 1, 1, 0, - 0, 0, 0, 0, 0, 0, 0, 0, 1, 1, 1, 1, 1, 1, 0, 0 -}; - -// first index is joined_stereo, second index is 0 or 2 (1 is unused) -static const float dequant_1bit[2][3] = { - {-0.920000f, 0.000000f, 0.920000f }, - {-0.890000f, 0.000000f, 0.890000f } -}; - -static const float type30_dequant[8] = { - -1.0f,-0.625f,-0.291666656732559f,0.0f, - 0.25f,0.5f,0.75f,1.0f, -}; - -static const float type34_delta[10] = { // FIXME: covers 8 entries.. - -1.0f,-0.60947573184967f,-0.333333343267441f,-0.138071194291115f,0.0f, - 0.138071194291115f,0.333333343267441f,0.60947573184967f,1.0f,0.0f, -}; - -} // End of namespace Audio - -#endif diff --git a/sound/module.mk b/sound/module.mk index cd1ff0df8e..df593d8e1f 100644 --- a/sound/module.mk +++ b/sound/module.mk @@ -18,7 +18,6 @@ MODULE_OBJS := \ decoders/flac.o \ decoders/iff_sound.o \ decoders/mp3.o \ - decoders/qdm2.o \ decoders/raw.o \ decoders/vag.o \ decoders/voc.o \ |